Re: SV: [Asterisk-Users] Nokia E61
Hi, I had no issues connecting/calling to my asterisk box (on public ip), even my phone is behind a hotspot. Its just that i need to use G711 codec. At 03:34 PM 7/5/2006, you wrote: Hello Has anyone tried a Nokia E6x phone when it is natted? Like behind a hotspot or similar? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Devraj Mukherjee Sendt: 4. juli 2006 12:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Nokia E61 Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? ___ --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] Nokia E61
Hi, configuration for E61 is the same as E60. As for the codec, G729 works between E60/61 phones (G729 passthru). At 03:44 PM 7/4/2006, you wrote: Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --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] Nokia E60 , experience as SIP client
try using g711 ulaw codec. At 03:51 PM 5/31/2006, you wrote: Hi I want to check out from the members , about their experience with Nokia E60 phone as SIP client , I was able to register the phone , but my voice gets broken during the calls . My other Wi-Fi VoIP SIP phone are working fine I also like to check out is there any other mobile manufacture who have SIP supported porducts like Nokia e-60 Thanks Joseph John ___ --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] Incoming Asterisk SIP DID Calls
Hi, Try this on your extensions.conf [from-mysipprovider] exten = 1140636249,1,Dial(SIP/29650) exten = 1140636249,2,Hangup At 07:34 AM 3/30/2006, you wrote: Hello All, I am using incoming DIDs for the first time. I ll very happy if someone can help me on that serttings ... I need to know how to answer calls from IP 200.123.123.1 with credentials abc123456:123456 and I d like to address to extention 29650 incoming calls from that number which is 1140636249. Also for out going calls I d like to use my own context as I use now. So I need to know how to add this incoming calls to extention 29650 keeping the existing out going dial plan Sip.conf register = abc123456:[EMAIL PROTECTED] [in-did] type=friend username=abc123456 fromuser=abc123456 secret=123456 host=200.123.123.1 fromdomain=200.123.123.1 context=from-mysipprovider port =5060 dtmfmode=rfc2833 disallow=all allow=g729 allow=g723 allow=ulaw Extensions.conf [from-mysipprovider] exten = 29650,1,Answer ; 29650 is the contact extension set on pap2 exten = 29650,2,Hangup Newton ___ --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 limit?
How about moving your voicemail users into db? At 03:50 AM 3/23/2006, you wrote: Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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 limit?
You can try using asterisk-addons http://www.voip-info.org/wiki/view/Asterisk+voicemail+database or asterisk realtime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail At 04:51 AM 3/23/2006, you wrote: Hi, Is there a howto to do this? I'm using voicemail.conf and sip.conf for my voicemail users. Does it really has a limit? Thanks, Ryan At 08:23 PM 3/23/2006, Antonio Rabena wrote: How about moving your voicemail users into db? At 03:50 AM 3/23/2006, you wrote: Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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] TE410P and SPANDSP
Hi, I also experienced broken page receiving fax on asterisk + spandsp with Digium TE410P. I also tried diff. versions of spandsp and asterisk, still no luck. I had no issues using the same asterisk + spandsp config with TE110P. Any ideas? At 09:21 AM 11/24/2005, you wrote: Hi, All Does any one has successful experience use te410p and spandsp together? Could they work well with all 120 channels receive/send fax at the same time? My practice is that rxfax always get broken fax page. Help! ___ --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] Asterisk + Voicetronix Card
HI, I'm using asterisk + voicetronix openswitch12 (using fxo). I just noticed when I call a pstn number (mobile number), asterisk will answer the call first before it actually dials the destination number. Is this normal? -- Executing Wait(SIP/192.168.1.130-081671b0, 1) in new stack -- Executing Dial(SIP/192.168.1.130-081671b0, vpb/1-12/911) in new stack == 1-12 requested, got: [vpb/1-12] == vpb/1-12: Calling 911 on vpb/1-12 == vpb/1-12: Dial parms for vpb/1-12 1/2000ms/4000ms/4000ms/12ms == vpb/1-12: Dial parms for vpb/1-12 tone 7-0 == vpb/1-12: Dial parms for vpb/1-12 tone 0-1 == vpb/1-12: Dial parms for vpb/1-12 tone 4-2 == vpb/1-12: Dial parms for vpb/1-12 tone 7-3 == vpb/1-12: Dial parms for vpb/1-12 tone 3-4 -- vpb/1-12: VPB Calling 911 [t=12] on vpb/1-12 returned 0 vpb/1-12: chanreads: starting thread -- Called 1-12/911 -- vpb/1-12 is ringing == vpb/1-12: Dialend -- vpb/1-12 answered SIP/192.168.1.130-081671b0 == vpb/1-12:Now listening for DTMF == vpb/1-12: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] vpb/1-12: vpb_write: Starting play mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] == vpb/1-12: Hangup requested == vpb/1-12: Ending record mode (1/yes) == vpb/1-12: Ending play mode on vpb/1-12 Regards, Antonio ___ --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] CallBack Retries
Hi, Is it possible to disable the retries in callback? Regards, Antonio ___ 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] new ATA box for sale by Linksys
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Matthew Boehm wrote: Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. Thanks, Matthew ___ 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] new ATA box for sale by Linksys
Yes there is.. the default setting is /init.cfg. Not sure about these parameters.. I cant find this provisioning setting on the user-guide. maybe anyone can help? Regards, Antonio Rabena Andres wrote: Antonio Rabena wrote: I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Can you confirm if under Advanced Settings there is a Provisioning Tab, and under it there is a space for a Profile Rule? ___ 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] Asterisk testbed for teaching connecting to a PRI-ISDN
You need an E1 back-to-back cable. Regards, antonio Francesco Delfino wrote: Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. I need the testbed to be representative of the real-world difficulties in putting on an Asterisk BOX for connecting to a PRI-ISDN: is other hardware needed (e.g. echo cancellers or failover switches)? Asterisk BOX (Simulate the Telco) with Digium TE405P | \ | E1 \ T1 | \ [What to put here?] [What to put here?] | \ | E1 \ T1 | \ Asterisk BOX (Company) Asterisk BOX (Company 2) with Digium TE405P with Digium TE405P Regards, Francesco Delfino ___ 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] Re: Send DTMF tone Like 'C' on connected call
Hi, How can i send dtmf tone upon connection? On Friday 30 January 2004 01:15, Nick Bachmann wrote: Dear to all someone know how is possible to have a DTMF tone like "C" AKA Alpha Tone (connect tone) to the caller? Yes, it's possible. Regards, Antonio
Re: [Asterisk-Users] Problem with EuroISDN E1
maybe you can if try this.. span=2,1,0,ccs,hdb3,crc4 Claus Futtrup wrote: Here you go. loadzone = no defaultzone = no span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 span=2,2,0,ccs,hdb3,crc4 bchan=32-46 bchan=48-62 dchan=47 Both E100P are connected to PSTN. Kind Regards Claus Futtrup ___ 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] hide caller id
try to put hidecallerid=no in your zapata.conf Pedro Vela wrote: Yes, my phone company has enabled the Caller ID hiden possibility, thats because with a Panasonic PBX works fine but with Asterisk not. Thanks for your aproach, what can I do now? Regards, Pedro ___ 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] Connecting PBX to Asterisk
Im trying to inter-connect my current PBX system and Asterisk. Asterisk has some users from different networks (internet).. I used cisco router using 4 fxs to pbx and SIP to asterisk. Is there any way i can allow the ip address of cisco to connect to my asterisk using SIP? IP Address of cisco is 192.168.0.254 here's a part of my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default srvlookup = yes pedantic = yes tos=lowdelay maxexpirey=360 defaultexpirey=120 disallow=all allow=ulaw allow=alaw [2101] type=friend context=sip-users secret= host=dynamic username=2101 qualify=yes nat=yes canreinvite=no and my extensions.conf [sip-users] exten =_21XX,1,Dial(SIP/[EMAIL PROTECTED]) [default] exten s,1,Hangup Regards, Antonio Rabena ___ 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] Zap Channels Hang
Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI show channels Channel (Context Extension Pri ) State Appl. Data Zap/31-1 (default 9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network 968290897 2 ) Ring Dial Zap/g2/68290897 Zap/30-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network 993841544 2 ) Ring Dial Zap/g2/93841544 Zap/29-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-1fa1 (network 996644687 2 ) Ring Dial Zap/g2/96644687 Zap/28-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-f3c0 (network 993871648 2 ) Ring Dial Zap/g2/93871648 Zap/27-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-aa22 (network 968627224 2 ) Ring Dial Zap/g2/68627224 Zap/26-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-e6e3 (network 965627780 2 ) Ring Dial Zap/g2/65627780 Zap/25-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-70b1 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/24-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-6e19 (network 963167838 2 ) Ring Dial Zap/g2/63167838 Zap/23-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-76ce (network 965699062 2 ) Ring Dial Zap/g2/65699062 Zap/22-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-12dd (network 965676388 2 ) Ring Dial Zap/g2/65676388 Zap/21-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-527d (network 962662272 2 ) Ring Dial Zap/g2/62662272 Zap/20-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/811586002-037a (default 964290118 2 ) Ring Dial Zap/g2/64290118 Zap/19-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-dc3c (network 965627640 2 ) Ring Dial Zap/g2/65627640 Zap/18-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1007-49ad (network 964255575 2 ) Ring Dial Zap/g2/64255575 Zap/17-1 (default s 1 ) Up Bridged Call SIP/1007-de63 SIP/1007-de63 (network 965699062 2 ) Up Dial Zap/g2/65699062 Regards, Antonio Rabena
Re: [Asterisk-Users] Zap Channels Hang
Im using the stable versoin 0.7.2. At 04:23 PM 4/1/2004, you wrote: What version of Asterisk are you using.. I updated to the latest CVS yesterday and have started having the same problem.. I am busy building a new box to use from my Asterisk so will see if it is still a problem and a fresh install.. later.. Regards, Antonio Rabena
Re: [Asterisk-Users] Zap Channels Hang
how about for the stable version? im using 0.7.2.. is there any known bugs in this release? should i upgrade to CVS? At 01:55 AM 4/2/2004, you wrote: On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote: Hi, I have same problem with zap channels. I have E100P installed on my asterisk box and I worked with CVS-02/22/04-16:30:20 and everything worked well (with Zap channels). I update asterisk to new cvs 2 days ago and incoming zap calls starts hanging. I have mgcp extensions defined in my extensions.conf and I see that if voicemail is enabled for extension and there are two concurent call (from Zap) to this extension, second call to voicemail are hanging in asterisk after user from Zap side hangs up. If there are no voicemail for extension the call are not hanging at all. May be these information will be helpfull to fix this bug. I noted the same problems with CVS from 03/30/2004 when incoming calls were sent to voicemail. Anyway I had to roll back to 03/05 since last Zaptel was giving me yellow alarms con my TE410P on a E1 PRI. -- Juanjo sin .sig Regards, Antonio Rabena
Re: [Asterisk-Users] two UA with the same usr/pwd
At 05:44 AM 2/18/2004, you wrote: 2. can Two SIP phones login to * at the same time with the same username/pwd ? how to prevent this? I also want to know if its possible to prevent multiple logins.. Regards, Antonio Rabena
[Asterisk-Users] multiple context in sip.conf
Hi all, Is it possible to have multiple context= for user configuration in sip.conf? Regards, Antonio Rabena ___ 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] multiple context in sip.conf
Hi, Is it possible to have multiple context= for user configuration in sip.conf? Regards, Antonio Rabena ___ 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
