similar to: CallerID Number incorrect in SIP packet

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

Is there a way to allow a user to dial an extension after listening to your voicemail instead of leaving a message? Example would be the big boss is on vacation and changes his out message to say "you can reach my assistant at by dialing 1234 now or leave me a message". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

I can not tell from the docs or from the mailing list what is the state of timezone support in the iCalendar package? If I want to parse an iCalendar file that has non utc dstarts and dends will it convert those times to UTC or otherwise allow me to do that? When I tried to parse an iCalendar input file started off with something like this: BEGIN:VCALENDAR METHOD:REQUEST

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

Are there any IP Providers that will pass Caller ID? Broadvoice used to but no they dont. THX

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

I have two dataframe columns of POXIXct data/times that include seconds. I got them into this format using for example zsort$ETA <- as.POSIXct(as.character(zsort$ETA), format="%m/%d/%Y %H:%M:%S") My problem is that when I subtract the two columns, sometimes the difference is given in seconds, and sometimes it is given in minutes. I don't care which it is, but I need to know which

Good afternoon! I'm trying to model a time series on the following data, which represent a monthly consumption of juices: >x<-scan() 1: 2859 3613 3930 5193 4523 3226 4280 3436 3235 3379 3517 6022 13: 4465 4604 5441 6575 6092 6607 6390 6150 6488 5912 6228 10196 25: 7612 7270 8617 9535 8449 8520 9148 8077 7824 7991 7660 12130 37: 9135 9512 9631 12642

Hi Arnaud, I received several replies from technical support. It turned out that according to the UPS serial number it had been manufactured in 2012. So, the device should support the ability to disable the beeper. But still I can not disable the beeper using the latest version of UPSMON Plus in Windows: http://www.pcm.ru/data/download/public/soft/setup_upsmon_v2.9.rar (don't know why; the

Hi All, Where can I find copies of the app_rxfax.c, app_txfax.c and apps_Makefile.patch. They don't seem to be located at soft-switch.org anymore. I am currently trying to compile Asterisk 1.2.26.1 and need the fax components. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial

Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? Thanks

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

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I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to turn this off? There is no extension defined for # in the dial plan. Thanks for your thoughts on this.

Does anyone know how to configure a SIP phone to pass the mailbox number to the voicemail service when dialing? I would like to press the message waiting lamp and be prompted for my password instead of "mailbox number". Can this be passed in the set-up call or based on caller-id? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I was wondering if the following was possible: When somebody sends a fax to my direct number 0101234567105 (my extension will be 105) is it possible that Asterisk, or an addon sees this as a fax, and e-mail the fax to me? So everybody with a private extension will be able to receive faxes in his e-mailbox on his direct number. Any pointers would be highly appreciated! Thanks, Peter