[Asterisk-Users] Forward calls from PSTN to PSTN very choppy
Hi all, I am playing with asterisk forwarding my calls to my cell phone. I have 2 X101P boards. When I call in, type the extension number that calls my cell #, all works as expected. Except, the audio is terrible. It is very choppy. I played with jitterbuffers, and that helped a little bit, but still the audio is very bad. I have echo problems on both lines, and havent had much success in getting rid of those. Could this be related? Any pointers? Thanks, Mike ___ 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] email notification not working anymore
It uses sendmail. Try sending a test message from the computer first, to rule out asterisk as a problem. --Mike On Thursday, June 5, 2003, at 01:05 PM, Derek Beaumont wrote: I used to have email notification working with my voicemail services but it stopped working when I installed the new version of asterisk. I have not changed my voicemail.conf file, so I'm out of ideas. Does asterisk use Sendmail to send messages, or does it have its own method for sending email? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OS X support?
Nope not unless you want to port it. --Mike On Thursday, April 3, 2003, at 02:17 AM, Roy Sigurd Karlsbakk wrote: hi Can I use Asterisk with OS X? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
For those of you running OSX, a new h323 client was released. Haven't set up h323 yet, so I can't vouch for it. http://xmeeting.sourceforge.net/ --Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How could I get * from CVS if I am not on the Linux platform?
WinCVS... www.wincvs.org. --Mike On Friday, March 28, 2003, at 05:09 PM, it wrote: Hi,I want to get the latest asterisk code from CVS. But the computer OS I used for travelling internet is Windows. I don't know how to I deal with the CVS. Thanks. john -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy.
Re: [Asterisk-Users] Dialing SIP
Replace exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED] with: exten = 301,1,Dial,SIP/301 --Mike On Wednesday, March 26, 2003, at 08:40 AM, Benjamin J. Bawkon wrote: Im really starting to get the hang of Asterisk, however, I still have one issue... My SIP Client can dial other extensions just fine, but no extension can ring the Sip client... Here is the pertinent info: SIP.CONF, [general] port = 5060 bindaddr = 192.168.0.5 ;ip of asterisk server context = default [301] username=301 context=local type=friend secret=test insecure=yes host=dynamic EXTENSIONS.CONF [local] exten = _1XX,1,Dial,ZAP/1/BYEXTENSION exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED]; again, ip of * server blah blah blah below this.. Console Debug: When 301 is Dialed: --Executing Dial(OSS/dsp, SIP/sip:[EMAIL PROTECTED]) in new stack Called sip:[EMAIL PROTECTED] Got SIP response 482 Loop Detected back from 192.168.0.5 No one is available to answer at this time WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no rule 't' in context 'local' Problem is, the SIP Client never rang. Now...If I change the extensions.conf to read: Exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED] Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address to the sip client machine...It will change occasionally... Any Ideas? Thanks! Ben Bawkon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #198 - 7 msgs
SIP/301 not SIP.301. --Mike On Wednesday, March 26, 2003, at 10:35 AM, Benjamin J. Bawkon wrote: Replacing that line as directed gives this as console debug: Executing DIAL(OSS/dsp, SIP.301) in new stack NOTICE[155663]: File app_dial.c, Line 449 (dial_exec): Unable to create channel of type 'SIP' Everyone is busy at this time WARNING[155663]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no rule 't' in context 'local' Could it be my sip client? I'm using SJphone currently. Subject: Re: [Asterisk-Users] Dialing SIP From: Mike Reiling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Replace exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED] with: exten = 301,1,Dial,SIP/301 --Mike On Wednesday, March 26, 2003, at 08:40 AM, Benjamin J. Bawkon wrote: Im really starting to get the hang of Asterisk, however, I still have one issue... My SIP Client can dial other extensions just fine, but no extension can ring the Sip client... Here is the pertinent info: SIP.CONF, [general] port = 5060 bindaddr = 192.168.0.5 ;ip of asterisk server context = default [301] username=301 context=local type=friend secret=test insecure=yes host=dynamic EXTENSIONS.CONF [local] exten = _1XX,1,Dial,ZAP/1/BYEXTENSION exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED]; again, ip of * server blah blah blah below this.. Console Debug: When 301 is Dialed: --Executing Dial(OSS/dsp, SIP/sip:[EMAIL PROTECTED]) in new stack Called sip:[EMAIL PROTECTED] Got SIP response 482 Loop Detected back from 192.168.0.5 No one is available to answer at this time WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no rule 't' in context 'local' Problem is, the SIP Client never rang. Now...If I change the extensions.conf to read: Exten = 301,1,Dial,SIP/sip:[EMAIL PROTECTED] Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address to the sip client machine...It will change occasionally... Any Ideas? Thanks! Ben Bawkon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialout Zap1/1
exten => 555,1,Dial(Zap/1) Change 555 to the exten you want, or change it to a pattern. --Mike On Wednesday, March 26, 2003, at 10:55 AM, Michael K. Rodriguez wrote: Any ideas on how to dialout exten => zap 1/1 image.tiff> Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen,TX78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++ -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy.
Re: [Asterisk-Users] DIAL - Sip Extentions
Hi Ben, Try Dial(SIP/ben) Make sure you have a sip.conf entry for ben, or whatever you want. Yes, each phone needs an entry. If you need a example, let me know. --Mike On Sunday, March 23, 2003, at 11:46 AM, Benjamin J. Bawkon wrote: Hi All, Im very new to Asterisk, so can someone give me the context to use In the Extentions.conf file to ring a SIP Phone?(Cisco 7960) ? Also, does each IP Phone (again, Cisco 7960) need a separate entry In the SIP.CONF File? ? Thanks a ton! Ben Bawkon ? P.S. a sample SIP.CONF file that works well for multiple DHCPd Cisco 7960s Would be really nice to be able to look over.
Re: [Asterisk-Users] Cisco 7960
I tried it last night. Loaded fine, but the phone had no soft keys and could not get a dialtone. I assume my CARD.XML file is not correct, but it is the one I got off Cisco's site. --Mike On Wednesday, March 19, 2003, at 10:02 PM, Stephen Webb wrote: Has anyone used the MGCP firmware image? I have only seen talk about the SIP image. Just curious! Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 (SIP) XML
I know this isn't really related to *, but I thought some of you might be able to shed a little light. I have a 7960 running SIP. No matter what I send the phone, I get CMXML parse errors. I have the services_url set to a perl app that I keep changing with no luck. I have set the correct content-type. Does anyone have example XML that I can look at. I don't have the cisco call manager. Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Echo
Hello everyone, I finally got my Cisco 7960. Seems like a very nice phone. I have it configured for SIP and working with asterisk. But, I am having some echo problems. The person I call does not hear an echo, but I hear my own voice echo back to me about .5 seconds later (maybe a little less). Does not matter if I call an internal Zap extension, or outbound over my X100P. I changed the codecs, around and the one that gives the least amount of echo is g729a. It is plugged in my office, but my asterisk server is in the attic, with a wireless link. Could that be a problem. There is very minimal latency. A traceroute from the phone shows this: SIP Phone traceroute cartman traceroute to cartman(192.168.100.3), 30 hops and 40 byte packets SIP Phone 1)192.168.100.3 10ms1ms 10ms The server is a celeron 433. Should I toss a faster processor at the computer, or is it the wireless link? I can't really image it being the wifi link, as VOIP works over the internet, with much higher latency. Any suggestions? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Echo
No, since the echo happens even when calling locally. Should I turn it on for the T100P (analog phones attached to channel bank)? Thank, Mike On Friday, March 14, 2003, at 12:18 PM, Richard Tomson II wrote: Have you turned on the echo cancellation on the X100P? Check the zaptel Makefile. I use the Mark2 with Aggressive_suppressor. Rich. On Fri, 2003-03-14 at 10:42, Mike Reiling wrote: Hello everyone, I finally got my Cisco 7960. Seems like a very nice phone. I have it configured for SIP and working with asterisk. But, I am having some echo problems. The person I call does not hear an echo, but I hear my own voice echo back to me about .5 seconds later (maybe a little less). Does not matter if I call an internal Zap extension, or outbound over my X100P. I changed the codecs, around and the one that gives the least amount of echo is g729a. It is plugged in my office, but my asterisk server is in the attic, with a wireless link. Could that be a problem. There is very minimal latency. A traceroute from the phone shows this: SIP Phone traceroute cartman traceroute to cartman(192.168.100.3), 30 hops and 40 byte packets SIP Phone 1)192.168.100.3 10ms1ms 10ms The server is a celeron 433. Should I toss a faster processor at the computer, or is it the wireless link? I can't really image it being the wifi link, as VOIP works over the internet, with much higher latency. Any suggestions? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
Morning, Actually the phone will be running SIP, but from what I have read, most people use the call manager for the services_url stuff. I read the url mentioned below, and wrote some simple xml, but the phone services emulator doesn't like it. My guess is that just doesn't work correctly. With all that in mind, once I get my phone (hopefully tomorrow), I am planning on writing some XML files to be sent to the phone. What things would people like to see developed. My first script will be for the directory, using voicemail.conf. Thanks, Mike On Wednesday, March 12, 2003, at 08:40 PM, Stephen Webb wrote: What mode are you running the Phone in? SIP, MCGP, or SCCP (Skinny) You mentioned Call Manager so I will assume SCCP. If that is the case I do not know. However if you are running it in SIP, All you have to do is set # XML URLs services_url: ; URL for external Phone Services directory_url: ; URL for external Directory location logo_url: ; URL for branding logo to be used on phone display These in you configuration and point it to a webserver. The xml format can be found here. http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/ bxtml.htm Hope this helps! Stephen On Wed, 2003-03-12 at 18:59, Mike Reiling wrote: Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960
Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call parking - Still haven't solved
Hi all, I got a lot of good suggestion on call parking not working, but I still haven't been able to make it work To recap, I have 701-720 showing up in the dialplan, and I can call those numbers just fine. But, if I try to hook-flash and transfer a caller to 700, it fails. I get a message that 700 is not a valid extension. I am including my extensions.conf and parking.conf below. Can anyone see anything obvious? Thanks, Mike --- parking.conf --- ; ; Sample Parking configuration ; [general] parkext = #700; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 60 ; Number of seconds a call can be parked for (default is 45 seconds) --- extensions.conf --- [general] static=yes writeprotect=yes [incoming] exten = s,1,DigitTimeout(5) exten = s,2,ResponseTimeout(10) exten = s,3,PrivacyManager exten = s,4,LookupBlacklist exten = s,5,LookupCIDName exten = s,6,Dial(Zap/1Zap/2Zap/3Zap/4Zap/5Zap/6Zap/7,18) exten = s,7,Answer exten = s,8,Wait(1) exten = s,9,BackGround(vm/1/unavail) exten = s,10,Voicemail(s1) exten = s,11,Hangup exten = s,104,Goto(privacy_man,s,1) exten = s,105,Goto(blacklisted,s,1) exten = i,1,Playback(invalid) exten = i,2,Goto(s,8) exten = 0,1,VoicemailMain(1) exten = 0,2,Playback(vm-goodbye) exten = 0,3,Hangup exten = fax,1,Dial(Zap/16) [blacklisted] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Zapateller exten = s,4,Playback(tt-allbusy) exten = s,5,Hangup [privacy_man] exten = s,1,Zapateller exten = s,2,BackGround(vm/1/unavail) exten = s,3,Voicemail(s1) exten = s,4,Hangup [extensions] exten = 10,1,Dial(Zap/1r2Zap/2r2Zap/3r2Zap/4r2Zap/5r2Zap/6r2Zap/7r2) exten = 11,1,Dial(Zap/1r2) exten = 12,1,Dial(Zap/2r2) exten = 13,1,Dial(Zap/3r2) exten = 14,1,Dial(Zap/4r2) exten = 15,1,Dial(Zap/5r2) exten = 16,1,Dial(Zap/6r2) exten = 17,1,Dial(Zap/7r2) exten = 80,1,VoicemailMain(s1) exten = 80,2,Hangup exten = s,1,DigitTimeout(5) exten = s,2,ResponseTimeout(10) exten = i,1,Playback(invalid) include = parkedcalls [unrestricted] exten = #,1,Goto(extensions,s,1) exten = 123,1,WakeUpMain exten = 123,2,Hangup exten = _1NXXNXX,1,Dial(Zap/g1/BYEXTENSION) exten = _1NXXNXX,2,Congestion exten = _NXX,1,Dial(Zap/g1/BYEXTENSION) exten = _NXX,2,Congestion exten = _011.,1,Dial(Zap/g1/BYEXTENSION) exten = _011.,2,Congestion exten = _N11,1,Dial(Zap/g1/BYEXTENSION) exten = _N11,2,Congestion exten = s,1,DigitTimeout(5) exten = s,2,ResponseTimeout(10) exten = t,1,Hangup exten = i,1,Playback(invalid) [restricted] exten = #,1,Goto(extensions,s,1) exten = s,1,ResponseTimeout(10) exten = t,1,Hangup exten = i,1,Playback(invalid) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call parking - Still haven't solved
Did that... Doesn't seem to help On Monday, March 10, 2003, at 09:49 AM, James Sharp wrote: parkext = #700; What ext. to dial to park Try removing the # -- Mike Reiling Systems Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slight Echo problems
Does anyone have the solution for this. I have a T100P going to a channel bank for all internal extensions. Calls from one internal to another sounds prefect. When placing a outbound call (going out over X100P) there is a very slight echo, almost like the comfort noise is delated by a fraction of a second. Its just a little annoyong, but when moving papers on my desk, it is very noticeable. I turned echo cancellation on for channels 25-26 (X100P), but that doesn't seem to do it. Does the gain affect this in any way? Would a faster computer help? Right now it is running on a 433Mhz celeron. Thanks everyone, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
