[Asterisk-Users] Voicemail as email attachment not working individually i.e. extensions specific
Hello I am using asterisk 1.0.0, here i am facing one problem that the email-aatchment setting for each extesion is not working individually. When globally attach=yes is set the voicemail will be sent as attachment no matter for any extension if attach=no is set for it. Same in the case with if attach=no is set globally then attach=yes will not work if set for some particular extension. I have googled for it but couldnt found anything useful.please help.. Thanks Asterisk-user __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Multicast
Hi, Could anyone tell me if asterisk supports multicast? And if so, what type? And if not, are there any plans to implement one in the forseeable future? Thanks, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceXML support and integration
Hi All, Do any of you know what the status is for VoiceXML support in * ? Is it already existing, or is it planned for the future? If it's not in now, do you know on what type of scale the work would be to integrate VXML into * ? Thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Video/H323/SIP
Florian Overkamp wrote: Hi, -Original Message- Is there any software based solution to establish a video connection with * and sip protocol? MSN messenger 4.7 with any windows capturing device should work. Make sure you force the codecs properly, because MSN tries to negotiate some form of MJPEG which Asterisk doesn't support. Best regards, Florian How do you force the codecs? Do you do this in Messenger or Asterisk? Right now I have set videosupport=yes and allowed h261 and h261 in sip.conf. Are there any other settings I need to change? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Windows Messenger+Video in *
Florian Overkamp wrote: Hi, -Original Message- Has anybody used Windows Messenger with asterisk? All documents around (google - wiki - bugs.digium.com) say that asterisk supports windows messenger with video but i have no succes yet! I can establish connection with audio but no video yet. I've used a range of windows messengers from version 4.7 to 5.0.0482. This is a little brief to say. I have had this working properly with recent asterisk boxes. A few things: Check if the [general] section has 'videosupport=yes' and if the sip peers are allowed to use h261 and h263 codecs. Best regards, Florian Do you think you could post your relevant .conf files? Is sip.conf the only one affected? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Windows Messenger Problem
Hi All, I'm currently having a problem with video calls with 2 Windows Messenger clients through *. The video+audio call gets established ok, but after a random period, usually anywhere from 20 seconds to 3 minutes, (though usually under a minute), both video and audio gets disconnected. Just plain audio calls between WM through * work fine and don't have this problem. Using asterisk's sip debug function I see that it's always the initiator of the call that sends an untimely BYE message that terminates the call. Though I can't figure out why it's doing so, and seemingly at random. in my [general] section in sip.conf I enabled videosupport=yes, and under the corresponding phones I enabled allow=h261 and allow=h263. I tried both canreinvite=yes and no, and both resulted in the same problem. Does anyone have any ideas? Thanks in Advance, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN -- X100P -- asterisk -- sipphones. Thanks Johan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no incoming pstn ring tone
Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN -- E100P -- asterisk -- sipphones. Thanks Johan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No read routine on channel AsyncGoto/Zap/1-1ZOMBIE
I saw the error: No read routine on channel AsyncGoto/Zap/1-1ZOMBIE in my log today. Despite googling, I have no idea what this error relates to. Could someone please help me. Thanks JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need an bench-marking tool
hi, i wanted to test a conference, so can any one help me in finding out a bench-marking tool in which we can set different codecs for each user. with regards vicky ___ --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] unable to call ATT audio conference bridge
Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ___ --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] [Fwd: asterisk-users Digest, Vol 26, Issue 166]
I tried by adding answer() to the dial plan but the problem still exists. I am not sure if I am doing this right. Attached is the log file from asterisk while making the call to the conf bridge after adding answer() Could you please let me know if you find anything out of this log file? thanks for your help. Original Message Subject:asterisk-users Digest, Vol 26, Issue 166 Date: Thu, 28 Sep 2006 07:42:43 -0700 (MST) From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Message: 19 Date: Thu, 28 Sep 2006 10:30:25 -0400 From: BJ Weschke [EMAIL PROTECTED] Subject: Re: [asterisk-users] unable to call ATT audio conference bridge To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote: Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ATT's IVR to collect the passcode is coming through as early media and since you haven't signaled to the phones that the phone is answered they're probably not letting you send DTMF through the bridge that isn't technically supposed to be there yet. Put an Answer() in your dial plan prior to sending the call out to the Dial() application to reach the bridge for these types of calls and this generally fixes your problems caused by someone else not signaling correctly. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ -- Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ' 208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'AGI' Sep 28 19:30:04 DEBUG[32330] app_queue.c: Device 'SIP/208' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 28 19:30:04 VERBOSE[32329] logger.c: recordingcheck|20060928-193004|1159486204.0: Outbound recording not enabled Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] db.c: Unable to find key '208/emergency_cid' in family 'DEVICE' Sep 28 19:30:04 DEBUG[32329] func_db.c: DB: DEVICE/208/emergency_cid not found
[asterisk-users] Asterisk Directory listing
How do I take out few extensions (vm enabled extensions) from the default company directory listing? thanks. ___ --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
Re: [asterisk-users] unable to call ATT audio conference bridge
Hello, Can someone help me with this please? Attached is the log file. thank you Original Message Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166] Date: Fri, 29 Sep 2006 10:31:21 -0400 From: asterisk-user [EMAIL PROTECTED] To: asterisk-users@lists.digium.com I tried by adding answer() to the dial plan but the problem still exists. I am not sure if I am doing this right. Attached is the log file from asterisk while making the call to the conf bridge after adding answer() Could you please let me know if you find anything out of this log file? thanks for your help. Original Message Subject:asterisk-users Digest, Vol 26, Issue 166 Date: Thu, 28 Sep 2006 07:42:43 -0700 (MST) From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Message: 19 Date: Thu, 28 Sep 2006 10:30:25 -0400 From: BJ Weschke [EMAIL PROTECTED] Subject: Re: [asterisk-users] unable to call ATT audio conference bridge To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote: Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ATT's IVR to collect the passcode is coming through as early media and since you haven't signaled to the phones that the phone is answered they're probably not letting you send DTMF through the bridge that isn't technically supposed to be there yet. Put an Answer() in your dial plan prior to sending the call out to the Dial() application to reach the bridge for these types of calls and this generally fixes your problems caused by someone else not signaling correctly. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ -- Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ' 208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '0' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'AGI' Sep 28 19:30:04 DEBUG[32330] app_queue.c: Device 'SIP/208' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 28 19:30:04 VERBOSE[32329] logger.c: recordingcheck|20060928-193004|1159486204.0: Outbound recording not enabled Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro' Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1' Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf' Sep 28 19:30:04 DEBUG[32329
[Asterisk-Users] Asterisk SIP cannot restrict call from softphone before registration
Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite.I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong password in the softphone. The registration and outgoing calls are failed as expected. Any experts encountered such problem before ? Please kindly help ! Million 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
[Asterisk-Users] Asterisk Realtime - How to enable the debug message for SIP users query
Hi experts, I wish someone would kindly give me a hand on a problem on Asterisk Realtime. May I know how to enable the debug messages for the Asterisk SIP Registrar query the SIP user data in the created MySQL table. I found that I can see the debug message for cdr_mysql which shows it can connect to Mysql successfully, but can't find any for app_addon_sql_mysql. Million thanks in advance ! 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] VoiceMail realtime not working in asterisk-1.2.6
hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail = odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer(SIP/xx.xx.xx.xxx-0a02e1c0, ) in new stack -- Executing Set(SIP/xx.xx.xxx-0a02e1c0, foo=102) in new stack -- Executing NoOp(SIP/xx.xx.xx.xxx-0a02e1c0, 102) in new stack -- Executing GotoIf(SIP/xx.xx.xx.xxx-0a02e1c0, 0?5:7) in new stack -- Goto (default,102,7) -- Executing VoiceMail(SIP/xx.xx.xx.xxx-0a02e1c0, 102) in new stack Apr 5 11:57:34 WARNING[14612]: app_voicemail.c:2385 leave_voicemail: No entry in voicemail config file for '102'. i also tried with adding searchcontexts=yes in voicemail.conf but i got segmentationfault. can any help me. with regards asteriskuser __ Yahoo! India Matrimony: Find your partner now. Go to http://yahoo.shaadi.com ___ --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
Re: [Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6
hi, odbc show is printing like Name: asterisk DSN: asterisk Connected: yes with regrads asteriskusers --- Nathan Bowyer [EMAIL PROTECTED] wrote: On 4/4/06, asterisk user [EMAIL PROTECTED] wrote: hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail = odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer(SIP/xx.xx.xx.xxx-0a02e1c0, ) in new stack -- Executing Set(SIP/xx.xx.xxx-0a02e1c0, foo=102) in new stack -- Executing NoOp(SIP/xx.xx.xx.xxx-0a02e1c0, 102) in new stack -- Executing GotoIf(SIP/xx.xx.xx.xxx-0a02e1c0, 0?5:7) in new stack -- Goto (default,102,7) -- Executing VoiceMail(SIP/xx.xx.xx.xxx-0a02e1c0, 102) in new stack Apr 5 11:57:34 WARNING[14612]: app_voicemail.c:2385 leave_voicemail: No entry in voicemail config file for '102'. i also tried with adding searchcontexts=yes in voicemail.conf but i got segmentationfault. What does odbc show give for output? I'm running 1.2.6 on a box and its working just fine, pulling mailbox information from Realtime. Nathan ___ --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 __ Yahoo! India Matrimony: Find your partner now. Go to http://yahoo.shaadi.com ___ --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] QSIG support in Asterisk
I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks ___ --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] QSIG support in Asterisk
I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks --dp ___ --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
Re: [Asterisk-Users] QSIG support in Asterisk
I am trying to use QSIG to interoperate with legacy PBXs. I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI works with QSIG support in Asterisk. Thanks in advance. --Pillai On 5/4/06, Olivier Krief [EMAIL PROTECTED] wrote: 2006/5/3, Marco Mouta [EMAIL PROTECTED]: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf I've made some tests using this in Portugal and seems to work:--- switchtype=qsig ; you may try this in your zapata.conf --I suppose you are using PRI access.But i've also red that qsig is not fully compliant in Asterisk. I've found some PRI states not recognized in asterisk sending CAUSECODE=98 to my legaccy PBX when it seems to me that Asterisk should handle it... Any ways it seems to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco Mouta For curiosity, what sort of benefit were you after using QSIG ? Most vendors tout SIP as interoperability protocol.Did you use QSIG as a way, for instance, to provide Message Waiting Indicator to digital phones ? Cheers___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] QSIG suopprt in Asterisk
I am trying to use QSIG to interoperate with legacy PBXs. I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI works with QSIG support in Asterisk. Thanks in advance. --dp ___ --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] SIP Registration Problem
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server: Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip:[EMAIL PROTECTED];tag=12e8dd0080754148To: sip:[EMAIL PROTECTED];tag=as2383b1dfCall-ID: [EMAIL PROTECTED]CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290Content-Length: 0 After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ? Thanks ___ --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] problem with registration of SIP phone
I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get this in the server:Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip: [EMAIL PROTECTED];tag=12e8dd0080754148To: sip:[EMAIL PROTECTED];tag=as2383b1df Call-ID: [EMAIL PROTECTED]CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290 Content-Length: 0After a few server restarts and/or phone restarts the phone registers ok.Any ideas why ?Thanks ___ --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] Re: problem with registration of SIP phone
I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries to re-register it fails. On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get this in the server:Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip: [EMAIL PROTECTED];tag=12e8dd0080754148To: sip: [EMAIL PROTECTED];tag=as2383b1df Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip: [EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290 Content-Length: 0After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ?Thanks ___ --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] Re: problem with registration of SIP phone
And one more update that may help to find a solution to this problem. If I run asterisk -rx reload the registration works fine until the next re-registration and then I have the same error again Is there some solution for this problem exept runnning asterisk -rx reload all the time ? On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries to re-register it fails. On 11/22/05, Asterisk User [EMAIL PROTECTED] wrote: I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get this in the server:Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip: [EMAIL PROTECTED];tag=12e8dd0080754148To: sip: [EMAIL PROTECTED];tag=as2383b1df Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip: [EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290 Content-Length: 0After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ?Thanks ___ --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] incoming caller doesn't hear rining.
Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 loadzone=uk defaultzone=uk zapata.conf [channels] usecallerid=yes language=en echocancel=yes echocancelwhenbridged=yes rxgain=-5% txgain=+5% immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn context=default group=1 callgroup=1 pickupgroup=1 channel = 1-8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys ZIP2
Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a Got SIP response 481 Call Leg/Transaction Does Not Exist back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any calls out from the phone, or calls into the phone. Occaisionally I get a busy tone when I try to dial also. This is the entry for the ZIP2 in sip.conf [phone10] type=friend username=phone10 host=dynamic dtmfmode=rfc2833 mailbox=110 context=sip Incidentally, when I place the ZIP2 in a local subnet 192.168.0.x, the web interface for the phone is quick. But when I place it on a public IP address, the web interface is all but unusable and times out 90% of the time. However when I ping the phone it comes back with a 1~5ms ping. Does anyone have any ideas? Has anyone got a ZIP2 to work wth Asterisk? Thanks in Advance, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura + Asterisk 1.2 + dtmf
Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits anywhere. What changed in version 1.2?? I've read many people with the same issue but with different phones, has anyone figure out what's wrong?? Oswald ___ --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
RE: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf
Hi, Thanks very much for this excellent explanation. I had played with it and the option that best worked for me was to set dtmf=INFO in the Sipura and dtmfmode=auto or dtmfmode=rfc2833 in sip.conf. I just could not fine the logic as to why I had to make it INFO. I hope Sipura/Lynksys/Cisco team come out with that solution soon. Thanks again, Oswald From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tracinet Sent: Monday, December 12, 2005 9:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf Oswald, I have had the same issues with Sipura devices since moving to Asterisk 1.2 as well. We use rfc2833 exclusively in our network and the Sipura devices just stopped working with regard to DTMF. After MANY packet captures comparing Sipura devices which did not work to Cisco devices that did work, it was found that the Sipura implementation of rfc2833 was not to spec (Sipura calls it AVT). Specifically, when the Sipura device sends the RTP packets for each DTMF digit, the mark bit is set to 1 for each packet instead of just the first packet's mark bit being set to 1. Previous versions of asterisk were not as strict with regard to rfc2833 which is probably why you did not have issues before. I have confirmed this issue with our Sipura vendor as well as some of the developers in the asterisk-dev IRC channel. Our vendor has taken all the packet captures and notified Linksys of this bug in hopes that updated firmware will be released soon as it appears to affect Sipura/Linksys phones and ATAs. In the meantime, there is a workaround that you can use to get *somewhat* accurate DTMF tones. If you set the device DTMF settings to INFO and you specify rfc2833 as the dtmfmode in sip.conf, the phone *should* pass DTMF digits as long as you are not using a speakerphone. If you are using a speakerphone, either pickup the handset when pressing tones or hit the mute button while pressing tones to avoid the tones getting duplicated by microphone pickup. Hopefully Linksys/Cisco/Sipura gets this fix out soon since the whole point of using rfc2833 for DTMF is to avoid getting duplicate and inaccurate tones sent as a result of microphone pickup and to pass the digits in their own RTP stream. Anyone on this list who is using Sipura devices and is having this AVT/rfc2833 DTMF issue, please contact your Sipura vendor and make them aware of this issue and ask them to notify Linksys (as Linksys will only deal with their resellers). Hopefully if enough noise is made, they will sense the urgency in getting this fixed and can put out a bug fix in the form of updated firmware. On 12/12/05, Asterisk User [EMAIL PROTECTED] wrote: Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits anywhere. What changed in version 1.2?? I've read many people with the same issue but with different phones, has anyone figure out what's wrong?? Oswald ___ --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 ___ --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
RE: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf
Thank you, that works. But for me INFO works better than AVT+INFO... what is the difference between them? Unfortunately Sipura's website lacks of a lot of information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, December 12, 2005 9:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf What does your sip.conf look like? I had this problem (Polycom Asterisk Sipura) after changing the settings on the Sipura, everything works fine now. I changed it to AVT+INFO. and in sip.conf I have rfc2833 On 12/12/05, Asterisk User [EMAIL PROTECTED] wrote: Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits anywhere. What changed in version 1.2?? I've read many people with the same issue but with different phones, has anyone figure out what's wrong?? Oswald ___ --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 ___ --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 ___ --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
RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
Has anyone tried out Hitachi IPC-5000 ? It looks nice and it's a bit exensive, but I would still like to hear how does it behave around Asterisk. Ivan ___ --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] Help setting IAX variables.
Hi All, I am new to Asterisk and want to perform following on my test project. I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB. Now I set some variables on SB in the same context where an IAX call lands. My question is , is it possible to access these variables in dialplan of SA? If yes then how? I know about IAXVAR application where variables set in source server of IAX channel can be access from destination server... Any help is greatly appreciated. ---Asterisk User ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help setting IAX variables.
Thanks Tilghman for your quick reply. I know that we should set variables through IAXVAR on source server to access them on Destination server. I just wanted to know the reverse case, where IAX channel variables set on destination server are accessible on Source server or not. Thanks again for your inputs. --- Asterisk user ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help sending call to local server
Hi,
I have a generalized syntax for dial application in my dialplan where I send
calls to particular server.
Here is my dial sysntax...
exten = _x.,1,Dial(${Dial_technology}/${extension_to_call}@
${Server_ip},30,r)
I can send a call to remote server using register statement in sip.conf or
iax.conf and it works as calls get landed in particular context of remote
server.
Would you please suggest me changes to be made in .conf file(s) if I want
the calls to be landed in context of local server if Server_ip is the IP of
a server running asterisk?
Thanking you
--ASTERISK USER
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[asterisk-users] help on ${RTPAUDIOQOS}
Hi All,
While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in
my dialplan.
I had 2 sip extensions 555 and 666 and I called from 555 to 666, but
unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
Would you please let me know what is wrong with my dialplan and/or what else
should be done to get the value of ${RTPAUDIOQOS}?
Following is my dialplan context where my call landed
[incoming_vpbx]
exten = _x.,1,NoOp(A call has come)
exten = _x.,n,Noop(${RTPAUDIOQOS})
exten = _x.,n,Dial(SIP/666,30,m)
exten = _x.,n,Hangup()
exten = h,1,Noop(***${RTPAUDIOQOS})
And here is what appeared on CLI...
-- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7a80948, A call has
come) in new stack
-- Executing [...@incoming_vpbx:2] NoOp(SIP/555-b7a80948,
) in new stack
-- Executing [...@incoming_vpbx:3] Dial(SIP/555-b7a80948,
SIP/666,30,m) in new stack
== Using SIP RTP CoS mark 5
-- Called 666
-- Started music on hold, class 'default', on SIP/555-b7a80948
-- SIP/666-089cb090 is ringing
-- SIP/666-089cb090 answered SIP/555-b7a80948
-- Stopped music on hold on SIP/555-b7a80948
-- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090
-- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7a80948,
***) in new stack
Thanking you...
---Asterisk User
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Re: [asterisk-users] help on ${RTPAUDIOQOS}
Klaus, Yes I do have set canreinvite=no in sip.conf. One more thing I noticed is following two cases when I replaced exten = _x.,n,Dial(SIP/666,30,m) with .exten = _x.,n,Dial(SIP/666,30,me) (1) When called extension(666) receives and hangs up the call. -- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7918e68, A call has come) in new stack -- Executing [...@incoming_vpbx:2] NoOp(SIP/555-b7918e68, ) in new stack -- Executing [...@incoming_vpbx:3] Dial(SIP/555-b7918e68, SIP/666,30,me) in new stack == Using SIP RTP CoS mark 5 -- Called 666 -- Started music on hold, class 'default', on SIP/555-b7918e68 -- SIP/666-09830108 is ringing -- SIP/666-09830108 answered SIP/555-b7918e68 -- Stopped music on hold on SIP/555-b7918e68 -- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108 -- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7918e68, **) in new stack -- Executing [...@incoming_vpbx:1] NoOp([1;35;40mSIP/666-09830108, **ssrc=1245221053;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00) in new stack == Spawn extension (incoming_vpbx, 12, 3) exited non-zero on 'SIP/555-b7918e68' (2)When called extension(666) receives and caller extension(555) hangs up the call. -- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7918e68, A call has come) in new stack -- Executing [...@incoming_vpbx:2] NoOp(SIP/555-b7918e68, ) in new stack -- Executing [...@incoming_vpbx:3] Dial(SIP/555-b7918e68, SIP/666,30,me) in new stack == Using SIP RTP CoS mark 5 -- Called 666:00*CLI -- Started music on hold, class 'default', on SIP/555-b7918e68 -- SIP/666-09830108 is ringing -- SIP/666-09830108 answered SIP/555-b7918e68 -- Stopped music on hold on SIP/555-b7918e68 -- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108 -- Started music on hold, class 'default', on SIP/555-b7918e68 -- Stopped music on hold on SIP/555-b7918e68 -- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7918e68, **ssrc=1405826681;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=101;rlp=0;rtt=0.00) in new stack -- Executing [...@incoming_vpbx:1] NoOp(SIP/666-09830108, **) in new stack == Spawn extension (incoming_vpbx, 12, 3) exited non-zero on 'SIP/555-b7918e68' So it looks like it has something to do with the way a call is hungup. Has anybody else any idea? Thanks, ---Asterisk User ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec translation in Asterisk
Hi Group, Can anybody explain me in detail how the codec translation happens on asterisk side when 2 endpoints have different codecs? Thanking you in advance. --SM -- _ -- 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
Re: [asterisk-users] Codec translation in Asterisk
Nobody to take this one! Am I missing anything in knowing following issue? --Hi Group, --Can anybody explain me in detail how the codec translation happens on --asterisk side when 2 endpoints have different codecs? --Thanking you in advance. SM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDILOOKUP doesn't return record
Hi All,
Found an issue with DUNDILOOKUP function in Asterisk 1.6.0.5.
I was using DUNDIQUERY (Set(ID=${DUNDIQUERY(${MNUM},priv,b)})) for
dundilookup and it was working fine.
But when I tried to use DUNDILOOKUP function
(Set(DL=${DUNDILOOKUP(${MNUM},priv,b)})), it didn't retuen me a
result. Moreover, the cli command 'dundi lookup 12...@priv' returned
me the result at the same time!
I also checked that ${MNUM} is set properly.
What can be a problem?
Please guide me where I do a mistake.
--SM
--
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[asterisk-users] Delay on sip channel
Hi, My SIP service provider terminates calls in meetme in my Asterisk PBX and am getting delay on those channels. I found following link to measure delay in meetme and to decrease it eventually. http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html It says, enable USE_RTC for dahdi_dummy. I have been using virtual server for hosting Asterisk and I had it disabled as per one had mentioned here to prevent the crashing which was happening earlier..(http://www.odindev.com/content/troubles-zaptel-centos-52-xen). Can you shed some light on the issue? --SM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF in Asterisk 1.4.*
Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF in Asterisk 1.4.*
Thanks Zeeshan, I have some problems with 1.4.25 so I'll try 1.4.27. Have a nice day! On Fri, Oct 29, 2010 at 1:33 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-29 5:15 AM, Asterisk User an.asterisk.u...@gmail.com wrote: Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan matching
Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten = _01137477.,3,Goto(intl-disabled,s,1) exten = _0113749.,4,Goto(intl-disabled,s,1) exten = _011.,5,Goto(intl-disabled,s,1) exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wellgate 3701
Hi everyone I'm trying to setup this Welltech Wellgate 3701 box. If I got to the proxy setup it seems to work but the Pstn incoming call always got a voice prompt from the Wellgate. Going to peer to peer mode seems to be better but I couldn't find any working configuration inside Asterisk. I do not really suffer from the registration problem because I doing all those trials with no password for the 3701 line configuration since I'm in a closed environment. Thanks for any help. Ml ___ Asterisk-Users mailing list Asterisk-Users@lists.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] FXO-FXS parameters
Hello, I'm trying to get feed back from other Asterisk users of Welltech WellGate 3701A / 3702A Or Micronet SP5012s / SP5014s Or Immix Tel C3-FXS/FXO Or Euro Teletech VIP-400 (All those are in fact the same product...) Trying to find/share ideas/comments about registrations issue, caller ID issue, Sip or H323, Peer to Peer or Gateway, Voice prompt on/off, Thanks Michel (FWD 627189) ___ Asterisk-Users mailing list Asterisk-Users@lists.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] CVS Changes (NAT-SIP)
I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Parking - Set ID on return
We have just analog lines coming in to our Asterisk box and so no CallerID information can be gathered, all calls look the same on the phone display. Once a user parks a call and the time runs out it returns the call but keeps the original CallerID information that makes it look like it is just another call from the outside. The operator has to go through the whole company greeting thing again before realizing it was a person who was just parked. Is there a way to set a new CallerID on that returned call so that the operator can skip the intro and go right to asking if they caller would like to go to voicemail instead? Thanks Phil Smith ___ --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] ZAP - Can't pickup calls on Analog Trunk
We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't even see anything happening on that channel. Analog lines are channel 1-4 the trunks are on channel 5 and 6. Both cards are Wildcard TDM400. Zapta.conf [channels] usecallerid=no busydetect=yes busycount=6 callerid=Outside Caller 555 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-1.5 txgain=0.0 musiconhold=default context=incoming group=1 signalling=fxs_ks channel = 1-6 Zaptel.conf loadzone=us defaultzone=us fxsks=1-6 What are we doing wrong? Phil Smith ___ --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] ZAP - Can't pickup calls on Analog Trunk
Are the trunks just pots lines (plain old telephone service lines)? If you don't know you could put an analog phone on the incoming lines and verify you can dial out. Also, if you call the line the phone should ring. If this is true then you will need fxoks in your pbx instead of the fxsks. They are lines from the phone company but unlike normal phone lines in the fact that they have multiple numbers per trunk line. Such as trunk 1 has the following numbers assigned to it: 555- 555-1112 555-1113 555-1114 555-1115 On the old PBX these lines are picked up and the then the called number is checked, from there the call is routed to the appropriate desk phone. Because of that I am not sure I can plug a standard phone in and get that phone to ring, I will give it a shot and see what happens. I can however make calls out of the trunk through Asterisk this leads me to believe that it is correct to have FXO ports using fxsks instead of FXS with fxoks. Phil Smith ___ --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] Voicemail retention
Asterisk version 1.2.27 We are running into issues where people are not deleting their voicemails and it is filling up the storage for voicemail. We would like to run a script that dumps all voicemail that are older than X days. Can we simply check the date time stamp on the message directory and delete those files older than X days or will that mess up the sequence of the voicemails? Anyone have a smooth way of doing this in 1.2? Thanks Phil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Show call coming back from Call Parking
Our operator has asked if it is possible that when a call times out in the call parking and comes back to her, if there is someway to show that call has come back from parking. I have looked all over the documentation and have come up with nothing so far. All I see when a call times out is: -- Stopped music on hold on Zap/25-1 == Timeout for Zap/25-1 parked on 702. Returning to park-dial,SIP/214,1 -- Executing Dial(Zap/25-1, SIP/214||t) in new stack -- Called 214 -- SIP/214-09086ff8 is ringing It appears that the park-dial is a context that Asterisk autogenerates so there is nothing I can do in that context. Has anyone else found a way to show that this a call returning and not a new call coming in? Thanks ___ --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
RE: [Asterisk-Users] Wengo config and G729(a)
Hello Here you go : [wengo-outgoing] type=peer fromuser= username username= username secret=password host=voip.wengo.fr fromdomain=voip.wengo.fr disallow=all allow=alaw allow=ulaw dtmfmode=inband canreinvite=yes nat=yes insecure=very dtmf=inband context=wengo-outgoing authname= username This is my current working config BUT to have it working, you have to add this entry to your /etc/host file (and reboot your Asterisk config...): 213.91.9.219voip.wengo.fr Hope this help Regards Michel LOPEZ [MVP Exchange] France __ -Message d'origine- De : [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] De la part de Remco Barende Envoyé : lundi 25 juillet 2005 19:45 À : [EMAIL PROTECTED] Objet : Re: [Asterisk-Users] Wengo config and G729(a) One way would do for me, I only use wengo for my outbound calls since they are a lot cheaper than our Royal Dutch KPN :) Which codec did you use and could you post your config lines? Thanks!! Remco On Mon, 25 Jul 2005, Wilson Pickett wrote: Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? I only have it working one-way, no incoming calls. Ironically, when Mark was here we caould have gone to meet them and straighten it out once and for all :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 / SIP tftp configs
I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cisco 7960 / SIP tftp configs
Thanks for the responses. All is happy. For the record the correct answers are: Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions do not. A1 - Don't just comment out the line setting, change it specifically to UNPROVISIONED. Q2 - How to get Message button working. A2 - Simply set messages_uri: where is the extension for VM. (Sorry but this should have been obvious, I did indeed find lots of stuff once I started searching on uri instead of url. Thanks for not burning me for not doing my research.) Note this line does not appear to be in the default SIPDefault.cnf file, you must add it manually. Q3 - How do I display an alias on the LCD for a registered line? A3 - In SIPx.cnf add line1_shortname: what I want displayed Note: this line does not appear in the default SIPxx.cnf file, you must add it manually. dbc. ___ --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] In France asterisk never detect hang up. Why ?
Hello. I am sorry my english is not good at all. When i have a call from a fxo port of a tdm400p, asterisk waits one minute before detecting that the caller has hang up his phone. I have in my extension conf : answer background (the prompt is 40 second long) dial (on fxs port) confgured for 30 seconds ringing. if the caller hang up at the begining of the background prompt, asterisk waits until he make ring the phone on the dial command for the all 30 secondes before detecting the hang up. Do you know if there is a way to repair that ? here is what i see on asterisk when the caller hang up IMMEDITALY after the test prompt begins : *CLI -- Starting simple switch on 'Zap/4-1' -- Executing Answer(Zap/4-1, ) in new stack -- Executing NoOp(Zap/4-1, 0675458745) in new stack -- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack -- Response timeout set to 20 -- Executing BackGround(Zap/4-1, barge) in new stack -- Playing 'test' (language 'fr') -- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1' -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' In my zapata.conf i have : language=fr default=fr relaxdtmf=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 cidsignalling=v23 usecallerid=yes group = 1 context=reseau signalling=fxs_ks callprogress=yes busydetect=yes callerid=asreceived busycount=5 pulse=yes In my zaptel.conf i have : loadzone=fr defaultzone=fr fxoks=1-3 fxsks=4 If anyone can see what is wrong he will really help me. thank you. ___ --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] Re: Asterisk en france
Bonjour, J'ai changé en tel que ci dessous, et j'ai toujours le même probleme. Il detect toujours pas le raccroché. I have changed to this new file, and i still have the same problem. Still not detecting hang up. [channels] language=fr default=fr relaxdtmf=yes rxgain=0.0 txgain=0.0 usecallerid=yes cadence=250,1500,1500,3000,1500,3000 echocancel=yes echocancelwhenbridged=yes echotraining=800 group = 1 context=reseau signalling=fxs_ks callprogress=no busydetect=yes callerid=asreceived busycount=3 pulse=yes channel = 4 group = 2 callgroup=2 pickupgroup=2 context=local signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes useincomingcalleridonzaptransfer=yes 2005/11/18, Dave Cotton [EMAIL PROTECTED]: Chez moi j'ai [channels] language=en callwaiting=yes callwaitingcallerid=yes callprogress=no busydetect=yes ;changed 17.03.04 from no busycount=7 ; added as above immediate=no usecallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 musiconhold=default ;faxdetect=incoming cadence=250,1500,1500,3000,1500,3000 Chez toi je me demande pourquoi cidsignalling=v23 callprogress=yes pulse=yes mes penses. -- Dave Cotton [EMAIL PROTECTED] ___ --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] cmd dial timeout don't work in asterisk 1.2 ?
Hello.
My dial timeout worked perfectly on the last asterisk but not on the new.
Here is my extension.conf :
exten = s,1,Answer()
exten = s,2,noop(${CALLERID})
exten = s,3,Set(TIMEOUT(response)=20)
exten = s,4,Background(test)
exten = s,5,Dial(Zap/2|${CALLERID},15)
exten = s,6,GoTo(personnedispo,s,1)
exten = s,106,GoTo(tousoccupe,s,1)
When it start to dial, it nevers stops and never go to 6 or 106.
Do you know why ?
Thank you for your help.
here is the output :
*CLI -- Starting simple switch on 'Zap/4-1'
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing NoOp(Zap/4-1, 0675123456) in new stack
-- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack
-- Response timeout set to 20
-- Executing BackGround(Zap/4-1, test) in new stack
-- Playing 'test' (language 'fr')
-- Executing Dial(Zap/4-1, Zap/2|0675123456|15) in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
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Re: [Asterisk-Users] cmd dial timeout don't work in asterisk
asterisk user dupont wrote:
Hello.
My dial timeout worked perfectly on the last asterisk but not on the new.
Here is my extension.conf :
exten = s,1,Answer()
exten = s,2,noop(${CALLERID})
exten = s,3,Set(TIMEOUT(response)=20)
exten = s,4,Background(test)
exten = s,5,Dial(Zap/2|${CALLERID},15)
exten = s,6,GoTo(personnedispo,s,1)
exten = s,106,GoTo(tousoccupe,s,1)
Ok problems with the above:
1) exten = s,5,Dial(Zap/2|${CALLERID},15) should not contain the | - that is
a separator for parameters, so you are setting a timeout of callerid and
options of 15.
So what do i have to use instead of | ?
As i am not in office today, i can not test.. but it seem curios to write :
exten = s,5,Dial(Zap/2${CALLERID},15)
no ?
I think i must use a separator ?
2) If you would like to have +101 bridging you need to use the j option to the
dial command now.
I didn't know that.
Thank you.
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RE: [Asterisk-Users] WiFi Phones
We bought a couple of the UTStarCom phones. They work fine in the office environment where noise is low, but on our production floor it is impossible for me to hear what is being said and the person on the other end of the call also says that they cannot hear a thing from the F1000 when the wireless phone is in a noisy environment. Still looking for a good WiFi phone for production/factory use. --Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Friday, October 07, 2005 5:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). [snip] The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Bought one from VoipSupply too, And yes, it doesn't support 802.1x radius auth (no place to select method, client certificate, etc). I've contacted voipsupply support about this and asked them to remove the 802.1x support listed on the product pages but got a cryptic reply that the phone does support 802.1x MD5.. (md5 is just a method of one of not supported 802.1x auths). Also, the max volume for the headpiece was actually quite low - in noisy environments as on streets you'll have hard time listening to the conversation. Overall, this phone is OK for home and small office use, nothing more. Just my $0.02 in, Vahan ___ --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 ___ --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
