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...digits, and if I call a SIP phone from the outside world, the beeps don't make it through. Here's what the packets look like, according to sip debug: Sip read: INFO sip:5854199708@208.34.86.35;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacyRPrQQu From:<sip:1017@m2k.netacc.net>;tag=1c11546 To: <sip:5854199708@208.34.86.35>;tag=as37f2b147 Call-ID: 2879790839083xQxj-1017--5854199708@208.34.86.37 CSeq: 3100834 INFO Supported: 100rel,em Content-Type: application/sdp Content-Length: 35 p=Digit-Collection y=Digits d=7 9 headers, 3 lines Receiving DTMF! Transmitt...

...g using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or anything at all involving SER and Asterisk? Thanks! -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi everybody, I want to read to debug messages and try to interpret them but they happen too fast, how can I log these guys to a file, or is there a file like this already? I checked the /var/log/asterisk but there isn't much interesting there yet? How can i turn on logging for SIP,IAX and other things? Thanks, Umut

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

...ork right all of the time. I don't forsee either involved telco doing anything at all about this. Is there any way to see if there is some sort of weird timing problem? Even better, is there a way to get things to work right on the TE410P? Thanks :-) -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

...0.wav -rwx------ 1 root root 64410 Oct 10 17:03 msg0010.WAV Note the gap between 0003 and 0009. This is caused by a somewhat common situation, and it tends to bite us somewhat often. :-) If not, if I get a chance, I'll whip something up. -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

...mail/etc. The Mediatrix doesn't answer (and therefore doesn't pass the call) until around the second ring, which is annoying, but them's the breaks. There's a bunch of other situations as well, but basically, it'll do most things. :-) -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net

Hi, I am translating the voice files of voicemail now. I don't know if the POUND and # are the same key in the telephone's keypad. If they are same, how could we understand the following message: %vm-msginstruct.gsm%To hear the next message press 6, to repeat this message press 5, to hear the previous message press 4, to delete or undelete this message press seven, to quite voicemail

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

Sorry if this is a resend, but it didn't appear to go the first time. > Sorry if this is not the correct place to post this..... > > > I have downloaded the cid_rewrite scripts that are located at: > http://www.muware.com/asterisk/ to my AAH v1.1 system. > > I apologize for my ignorance, but it says that I need to modify the > agi_config.php, but doesn't

Hello, I am getting a bizzare amount of error messages in the log files. The system seems to be running fine...no one is reporting any issues and all calls are coming and going. System is showing higher than average memory usage. eth0 is showing a high number of errors Running v1.1 Has happened on older versions and have been seeing this for quite some time but have just now asked if anyone