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...ndants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030623/d4b0cde3/attachment.htm

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=... context=interno ;auth=plaintext disallow=all allow=gsm allow=ulaw allow=alaw Any hint? I'm using a cvs from 4 days ago. PauloHM

...iginal Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com> Sent: Friday, October 17, 2003 12:23 PM Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk > Hi All, > > We've been developing for a while an IDE for Asterisk, and the time has > come to...

...middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress be of any help, as I'm outside the US? Thanks! PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/223f2550/attachment.htm

...or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM

...our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm

Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM

...users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=... context=interno ;auth=plaintext disallow=all allow=gsm allow=ulaw allow=alaw Any hint? I'm using a cvs from 4 days ago. PauloHM __________...

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

...stalling my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 E&m=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030812/cc74b8b5/attachment.htm

...15, I start getting a "multiframe alignment error" on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 e&m=1-15 2) When the test equipment tries to send me a DTMF string, I only get the first one. Any thoughts? Best, PauloHM

...to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a master reset? PauloHM

...n, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM -------------- next part -------------- A non-text attachment was scrubbed... Name: sipcallwaiting.diff Type: application/octet-stream Size: 3163 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030917/b105ef04/sipcallwaiting.obj

...'m working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM

...; error. I tried to find anything related at bugs.digium.com but couldn?t find any mention to this specific situation. My CVS version is quite old, but I would rather update it by a specific patch than to replace it entirely by a new one. Any hint would be greatly appreciated. Best regards, PauloHM

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hi folks, There was a bug with the GotoIfTime built-in command, under certain circumstances a variable contained garbage, screwing up correct time identification. I'm submitting now a patch to Mark so this can be fixed. PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030701/8f379910/attachment.htm

I have a client which is willing to connect * with a siemens hicon PBX. Any experiences that you wish to share, including FX, ISDN or E1 connectivity issues? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030707/dfb33d1f/attachment.htm

Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030730/0b768f8b/attachment.htm