Re: [asterisk-users] Strange behavior over Zap chennels
Hi, I was wondering if anyone you got asterisk to work with Fax via google voice ? If so, can you please send me extension.conf and sip.conf, jabber.conf and gtalk.conf settings used. I would prefer faxing with Fax for Asterisk (FFA) via .call file. I see post where people got it work with Google Voice. Guide is greatly appreciated. -Charles Fax Free http://sendfreefax.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk out of Media Path - Call Park
Hi all, Can i make Asterisk stay out of the media path forcall park feature?In the 'sip.conf' i made canreinvite=yes in the general sectionbut it does not seem to take effect. I don't see any reason for Asterisk to withhold sending re-invite. I am testing the call park in the single LAN,both on caller side and reciever side i am using X-Lite phones. Any suggestions?? Thanks, Sharath Chandra ___ --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] Oneway Audio
Hi all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902 phones and using cisco 2800 router. Codec is g711ulaw regards, -- Executing ParkedCall(SIP/192.168.50.2-09cbd610, 366) -- Channel SIP/192.168.50.2-09cbd610 connected to parked call 366Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) ___ --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] WARNINGS For SIP call
I am getting the following warnings on the Asterisk when i try a call parking scenario. I use Ciso 7920 phones and Cisco2800 Executing ParkedCall(SIP/192.168.50.2-088cde00, 366) in new stack -- Channel SIP/192.168.50.2-088cde00 connected to parked call 366Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) ___ --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] On ParkAndAnnounce and parking lot
I am using ParkAndAnnounce to Park the call and explicitly retrieving using ParkedCall app in the dial plan. I am trying to guess the parking lot being used in a particular call by incrementing a counter just before the ParkAndAnnounce and decrement the counter just before the ParkedCall. I am not sure if this is the right way to do. What i want to know is when is the parking lot released for recycling. Is is a safe assumption to decrement just beforeParkedCall. Thanks, Sharath ___ --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] Dial Out IVR
How can i configure the following scenario, - User 'A' dials into Asterisk, - Asterisk puts user 'A' on hold - Dials Out to User 'B' - Consults user B' if he wants to take the call (Press 1)or divert to voicemail (press 2) - Depending on the option chosen,either user A' call is bridged with the out call or transfered to voicemail. Thanks, Sharath Chandra ___ --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] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
Thanks Moj. But i need to connect to MySQL. Could this be a problemwith C libraries that i am using. Regards, Sharath On 3/8/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: This may not be the applicable solution, but if you're not using themysql config capabilities, add noload = res_config_mysql.so to modules.confMojSharath Chandra wrote: Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup'[res_config_mysql.so]Mar6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Mar6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! End=== Can someone suggest a solution. Regards, Sharath Chandra ___ --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 --Mojo [EMAIL PROTECTED]Office Manger, Horan Company, LLC(907) 747- x112___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup'[res_config_mysql.so]Mar 6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Mar 6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed!End=== Can someone suggest a solution. Regards, Sharath Chandra ___ --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: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
http://www.paesys.com/en/index.htm for the english versionOn 10/01/06, Guillaume de Lafontaine [EMAIL PROTECTED] wrote:HiI just discovered an interesting product line. Not tested yet... http://www.paesys.com/fr/lecteurs_VoIP_WiVoip_VideoIP_GSM.htmIn french, sorry...Any feedback ?--- Guillaume de Lafontaine___ D W A M ___ -Original Message- From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *? Sent: mar., 10 janv. 2006 13:17:02 GMT Received: mar., 10 janv. 2006 13:18:11 GMT Read: mar., 10 janv. 2006 13:22:28 GMT Rupert Gregory a écrit : Once you've finished drooling over the UTStarcom you can start drooling over the Linksys WIP330 http://ces.engadget.com/2006/01/07/linksys-wip330-in-da-house-but-you-cant-have-one/ VERY nice phone in my opinion. I dunno... it looks like a cell phone, except it's not one. It would be nice if it was a dual GSM / wifi phones which transparently switch to VoIP when you have a strong enough signal. This way, it would provide all the cost savings of VoIP with the convenience of GSM calls. Of course it would need to display a big bright icon to let the user know when they are not on wifi / voip since GSM providers are pretty expensive... Also, voicemail would become very nice. Get out of the office, the SIP register times out, and you're on voicemail. (of course you could also forward the call to the GSM number although it might be a little more expensive). X is out of the office kind of message would actually make sense... Get back in the office, the phone registers, and you get MWI. Cheers, Jean-Michel. ___ --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 listTo 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] zyxel p2000w
Hi Chip. I am absolutely certain that the P2000w does not have a call waiting feature. You may want to check the PBX, you may have to enter in a specific key - lie * or # - in order to answer the 2nd call. We use # for a blind transfer on our asterisk - asterisk picks up the P2000w tone no problem. hope it helps thanks Chandra Mistry On 21/11/05, cp [EMAIL PROTECTED] wrote: Does anyone know is the zyxel p2000w has call waiting? I hear noise when a second call comes in but cannot find any documentation. Thanks, Chip ___ --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 ___ --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] Help needed on setting up realtime
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database. Please suggest. Thank you. Sharath ___ 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] Basic telephony hardware questions
Hi, I am in the process of setting up an Asterisk-based PBX at work. I get the concept of how Asterisk works pretty decently. I am more confused about the proliferation of TLAs like FXO, FXS, TDP, SIP, After some intense reading I have come to some understanding of the hardware I need to set things up. I am doing this in India and am getting a friend of mine bring the cards with him when he comes here in a couple of weeks. I don't know when my next trip to a place of abundant Asterisk-capable hardware will be. So I cannot afford making the wrong decisions. I need to live with my choice for atleast the next 4-6 months. My Setup Our office currently has 3 (to expand into 4 in the next few months) incoming PSTN lines. All these lines have RJ11 terminations. There are a couple of hundred employees working in the office. But not all of them need or have a phone on their desktops. My Requirements --- I need about 10-12 POTS phones to be connected to the PBX. When their extension is dialled, they need to ring. I need about 3 IP-phones to connect to the PBX over Ethernet. There will be some 50 users who will use soft phones on their desktops to connect to the PBX to make and receive calls. I also need IVRS for incoming calls and voicemail for all the extensions. Based on all of the above: * The cheapest option for me to get started seems to be 4 Digium PCI cards on a box running Asterisk. Will a setup with an Asterisk box with 4 Digium cards work? * I have identified 'TDM04B - 4-port FXO bundle' as the card I need to connect to the incoming PSTN lines. Is this identification correct? Also, will the incoming RJ11 terminations connect to this card? Or do I need something else? * I have identified 'TDM40B - 4-port FXS bundle' as the card I need to connect my in-office POTS phones. Is this identification correct? Also, will these cards enable the connection of RJ11 cables connecting to the POTS phones? * What do I need to connect to my local 100-base-T LAN to the PBX? I want desktops on the LAN to be be able to run soft phones and connect to the PBX. Do I need any other card? * Similarly, do I need anything extra on the Asterisk box to connect the IP phones? * What are TE410 and TE405 cards used for? Any help will be appreciated. Thaths -- Lisa: Why are you dedicating your life to blasphemy? Homer: Don't worry, sweetheart. If I'm wrong, I'll recant on my deathbed. Slacker Without Bordershttp://openscroll.org/ Key fingerprint = 8A 84 2E 67 10 9A 64 03 24 38 B6 AB 1B 6E 8C E4 ___ 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] Cisco 2800 with Asterisk
Hi, Has anyone used Cisco 2800 Integrated services router to intiate SIP call to Asterisk. I would like to use it as gateway on to which T1 terminates and make Asterisk as my session target for few lines. Please let me know if there are any issues. Thanks, Sharath ___ 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] Asterisk with Softswitch
Hi, I am new to Asterisk. Can i use Asterisk as a session target from softswitch/Call Agent. I mean, is it possible to initiate a SIP call to Asterisk. My PRI terminates onto Cisco 2800 and i want to send few numbers to Asterisk to do some application related call control. Please advice ___ 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] grandstream asterisk configuration
i just saw a UDP blocked message in my gs GUI. ater i rebooted again i got MAC Address:00.0B.82.00.3C.13 Software Version:Program--1.0.3.81Bootloader--1.0.0.7 HTML--1.0.0.18 detected firewall/NAT type is open Internet assigning a STUN server also didn't help. lloked at the voip-info stuff a.. use dtmfmode=info in your sip.conf for your Grandstream BudgeTone and configure the GS accordingly b.. make sure to have a username=xxx entry in sip.conf that matches the phone's name as given in the square brackets c.. For most installations, this is needed in the sip.conf user definition (not in [general]): disallow=all allow=ulaw allow=alaw and did the same. still didn't work. what can be done if my nat is actually blocking the udp packets?? chandra - Original Message - From: SW [EMAIL PROTECTED] To: Chandra [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 10:30 PM Subject: Re: [Asterisk-Users] grandstream asterisk configuration Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: [EMAIL PROTECTED] i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra ___ 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 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] grandstream asterisk configuration
hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. chandra - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 9:50 AM Subject: Re: [Asterisk-Users] grandstream asterisk configuration On Wednesday 14 January 2004 01:52 pm, Mike Machado wrote: On Wed, 2004-01-14 at 08:45, SW wrote: Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). You don't need stun to work with Grandstream. My * is behind NAT and so is the GS of course. Two ports are open and redirected in the F/W, udp 4569 and 5036. I make and receive internal and external calls over both PSTN and the Internet. GS is configured: Software V 1.0.4.30 Static IP SIP Server is Asterisk's IP SIP user ID is the extension of GS Authenticate ID as user ID No pw Name is Steve Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723 723 Rate is 6.3 Silence Suppression is Yes Voice Frames are 2 IP SoQ is 48 VLAN 0 SIP User is NOT phone number Dial Plan 202 SIP register YEs Clear Reg oin reboot NO Expiration 60 Early Dial No Use # as Dial Key is Yes SIP port 5060 RTP 5004 Random port is No NAT traversal is NO keel alive is 20 TFTP server is 130.94.123.253 Voice mail ID is 78202 DTMF is in-audio Payload is 101 - this may need to be changed NTP time.nist.gov Now all my features used to work a few months ago. I then stopped using * and came back a week ago. Updated CVS and now Hold is not working unless I press #(!?) But I can call, receive, transfer and have a working V/M. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ 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 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] Asterisk (outside NAT) + BudgeTone (behind NAT)
I have been really trying to solve the this problem. Has anyone had a success on this one? I have asterisk setup outside my NAT with public IP and I am trying to establish a connection from Budgetone behind NAT with private IP. Everything seems to be working fine. They are registered, call rings successfully but there is a problem after the caller picks up the phone. IN CLI i constantly get: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable i also get some grandstream1 is now too lagged and after sometime i get grandstream1 is now reacheable messages.. i have this in sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT tos=lowdelay disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference dtmfmode=info [grandstream1] type=friend host=dynamic secret=grandstream1 context=outgoing nat=yes reinvite=no canreinvite=no qualify=200 help. ___ 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 + BudgeTone (behind NAT)
have u had any luck with this? cm - Original Message - From: Owen Kelso [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 12, 2004 9:51 AM Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT) Thanks for all of your responses. I confirmed that the phone works perfectly without NAT or through a IPSec VPN (yeah, I know, same thing). I've concluded that the Netgear router (FVS318) performing the NAT is corrupting the outgoing RTP packets. Traces confirmed that the BudgeTone is sending them out with a UDP checksum of 0 but the next hop after the Netgear router they are set to a non-zero value (an incorrect one). Asterisk is never even seeing the packets because the kernel is recognizing them as corrupt and dropping them, hence the recvfrom() Resource temporarily unavailable errors in rtp.c. I'm going to write Netgear to see what they have to say about it. If I make any progress I'll post to the list...thanks again, Owen ___ 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 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] grandstream asterisk configuration
hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is =i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:Resource temporarily unavailable my sip.conf file is as follows: [general]port =3D 5060 ; Port to bind tobindaddr =3D 0.0.0.0 ; Address to bind to;externip =3D 200.201.202.203 ; Address that we're going to put in =SIPmessages if we're behind a NATtos=3Dlowdelaydisallow=3Dall ; Disallow all codecsallow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1]type=3Dfriendhost=3Ddynamicsecret=3Dmysecretcontext=3Doutgoingnat=3Dyesreinvite=3Dnocanreinvite=3Dnoqualify=3D2000 has anyone done this before? chandra
Re: [Asterisk-Users] grandstream asterisk configuration
i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra - Original Message - From: bam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 6:42 PM Subject: Re: [Asterisk-Users] grandstream asterisk configuration Make sure that udp packets can get from the server back to the grandstream. At 12:40 14/01/04, you wrote: hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: [general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra ___ 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 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] Voicepulse
same here... with nufone too... i was just getting everyone is busy at the moment message in CLI... it was working fine before.. was it them or was something wrong with my network? will check tomm. cm - Original Message - From: Burak Balasaygun [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 8:21 PM Subject: [Asterisk-Users] Voicepulse I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ 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 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] sip and x-lite
try this... http://www.fnords.org/~eric/asterisk/ cm - Original Message - From: Ing Isianto Istiadi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 12, 2004 7:50 AM Subject: [Asterisk-Users] sip and x-lite Dear all, Can you give me the configurations for x-lite and sip in *. Thanks ___ 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 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 + BudgeTone (behind NAT)
can u give me the configuration for the firewall?? with the same configuration i can't even talk or hear... its giving me the RTP Read Error whenever one picks up the phone. cm - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 12, 2004 6:09 AM Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT) On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote: I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from me. From the console sip debug shows that the SIP part is working fine and DTMF via SIP INFO works. I use OpenBSD firewalls with NAT and redirect and it works just as it's supposed to. That's not even half duplex. In half duplex each side Can talk, but only one at a time. It seems to be an error with configuring your firewall. (One common error is to only turn on redirect. But you also need to Allow the traffic to flow... -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ 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 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 + BudgeTone (behind NAT)
i also had the same problem temporarily i solved my problem with both outside NAT. u can also do it if both inside NAT. * outside NAT and Budgetone behind NAT simply doesn't seem to work. if u ever solve this problem please let me know too. thanks cm - Original Message - From: Owen Kelso [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 11, 2004 4:52 AM Subject: [Asterisk-Users] Asterisk + BudgeTone (behind NAT) I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from me. From the console sip debug shows that the SIP part is working fine and DTMF via SIP INFO works. I've struggled with this for a few days now and can't figure out the cause. The only symptoms I've found are: (1) When I make a call the console spits out the following errors several times per minute: WARNING[-1220854864]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable (2) An ethereal trace reveals that incoming RTP packets have failed UDP checksums (all packets have the same checksum of 0xb38f). I don't see anything else irregular, like unreachable ports. My sip.conf contains: [test] type=friend username=test secret=12345 host=dynamic nat=yes qualify=1000 dtmfmode=info disallow=all allow=ulaw allow=alaw canreinvite=no On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060 for SIP and 19000 for RTP. The firewall that performs NAT forwards ports 5060 and 19000-19100 UDP to the phone. An ethereal snapshot looks like: 1.1.1.1 = Asterisk server 2.2.2.2 = Public IP where the BudgeTone is 10.0.3.205 = Private IP of BudgeTone Frame 211 (214 bytes on wire, 214 bytes captured) Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22 Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22) Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2) Type: IP (0x0800) Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2 (2.2.2.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) Total Length: 200 Identification: 0x (0) Flags: 0x04 Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x2538 (correct) Source: 1.1.1.1 (1.1.1.1) Destination: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: 13364 (13364), Dst Port: 19000 (19000) Source port: 13364 (13364) Destination port: 19000 (19000) Length: 180 Checksum: 0xdf43 (correct) Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) ..0. = Padding: False ...0 = Extension: False = Contributing source identifiers count: 0 0... = Marker: False .000 1000 = Payload type: ITU-T G.711 PCMA (8) Sequence number: 45554 Timestamp: 16480 Synchronization Source identifier: 1847249288 Payload: E4E4E5FAF9FDF0F6F5C2C5DFD0575D58... Frame 212 (214 bytes on wire, 214 bytes captured) Ethernet II, Src: 00:01:c7:0b:70:22, Dst: 00:06:29:ce:5f:f2 Destination: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2) Source: 00:01:c7:0b:70:22 (Cisco_0b:70:22) Type: IP (0x0800) Internet Protocol, Src Addr: 2.2.2.2 (2.2.2.2), Dst Addr: 1.1.1.1 (1.1.1.1) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) Total Length: 200 Identification: 0xe398 (58264) Flags: 0x00 Fragment offset: 0 Time to live: 233 Protocol: UDP (0x11) Header checksum: 0xd89e (correct) Source: 2.2.2.2 (2.2.2.2) Destination: 1.1.1.1 (1.1.1.1) User Datagram Protocol, Src Port: 19000 (19000), Dst Port: 13364 (13364) Source port: 19000 (19000) Destination port: 13364 (13364) Length: 180 Checksum: 0xb38f (incorrect, should be 0x1dc4) Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) ..0. = Padding: False ...0 = Extension: False = Contributing source identifiers count: 0 0... = Marker: False .000 1000 = Payload type: ITU-T G.711 PCMA (8) Sequence number: 53058 Timestamp: 3449661727 Synchronization Source identifier: 3820906983 Payload: D4D4D5D5D555D5D555D4D5D5D5D4D4D4... Frame 213 (214 bytes on wire, 214 bytes captured) Ethernet II, Src: 00:06:29:ce:5f:f2, Dst: 00:01:c7:0b:70:22 Destination: 00:01:c7:0b:70:22 (Cisco_0b:70:22) Source: 00:06:29:ce:5f:f2 (Ibm_ce:5f:f2) Type: IP (0x0800) Internet Protocol, Src Addr: 1.1.1.1 (1.1.1.1), Dst Addr: 2.2.2.2 (2.2.2.2) Version: 4 Header length: 20 bytes Differentiated Services
Re: [Asterisk-Users] asterisk sip with voicemail
i guess the user * looks for is the text within [] so i suppose the username and the text within [] should be same. try putting [nick] in place of [person]. actually, u don't need username as it only looks for the text between [] as username. cm - Original Message - From: Nick Knight To: [EMAIL PROTECTED] Sent: Friday, January 09, 2004 4:56 PM Subject: [Asterisk-Users] asterisk sip with voicemail Hello all, I have setup my sip.conf so users can register etc in the following format, [person] type=friend username=nick secret= host=dynamic mailbox=101 in my voicemail.conf I have an entry like 101 = 1234,Nick Knight,[EMAIL PROTECTED] Leaving a voicemail works fine after I have my dial command time out but on sip clients which display whether voicemail is waiting or not, it always displays No Voicemail (pingtel expressa). It emails them fine and I can call into the voicemail app which says I have new voicemail but I would like to resolve the sip client issue. Thanks Nick
[Asterisk-Users] GrandStream giving an RTP Read Error Again
- Original Message - From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 10:12 AM Subject: GrandStream giving an RTP Read Error hello, I have the following configuration now Asterisk(publicIP)-Netgear-Sip(privateIP) everything is registered ok. when Sip (grandstream-budgettone 100) calls the phones outside of netgear...suchas the zap phones just out side the netgear...or to external numbers... the rings goes ok.. but wwhen the other end picks up the phone, no voices are heard and in the CLI I can see a huge scrolling of this error: WARNING[20496]: File rtp.c, Line 375 (ast_rtp_read): RTP Read Error: Resource temporarily unavailable I have the following config for my grandstream [grandstream2] type=friend host=dynamic secret=grandstream2 context=outgoing nat=yes reinvite=no canreinvite=no qualify=60 and in my grandstream config i have dtmfmode=rtp2833. same for the sip.conf. can anyone help me out solve this problem? Thanks Chandra ___ 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] GrandStream giving an RTP Read Error
hello, I have the following configuration now Asterisk(publicIP)-Netgear-Sip(privateIP) everything is registered ok. when Sip (grandstream-budgettone 100) calls the phones outside of netgear...suchas the zap phones just out side the netgear...or to external numbers... the rings goes ok.. but wwhen the other end picks up the phone, no voices are heard and in the CLI I can see a huge scrolling of this error: WARNING[20496]: File rtp.c, Line 375 (ast_rtp_read): RTP Read Error: Resource temporarily unavailable I have the following config for my grandstream [grandstream2] type=friend host=dynamic secret=grandstream2 context=outgoing nat=yes reinvite=no canreinvite=no qualify=60 and in my grandstream config i have dtmfmode=rtp2833. same for the sip.conf. can anyone help me out solve this problem? Thanks Chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this script
has only first result although i have many in my database. this is the part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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Re: [Asterisk-Users] no results.
ok i guess,, we also have to put
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'account', '1234', '0');
at the beginning.. its working now
thankx
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 10:27 AM
Subject: RE: [Asterisk-Users] no results.
the database schema is the table and it's associated columns. did you use
the create table script in the header of the pl file? basically, for each
of your sip entries, they would be broken down per line. so if your
sip.conf entry looks like this:
[1234]
type=friend
username=1234
secret=blah
nat=yes
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'type', 'friend', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'username', '1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to disable an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have
something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into
the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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Re: [Asterisk-Users] no results.
ok thats done as u said. i am getting
No sip accounts defined in sip
error now.
??
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 10:27 AM
Subject: RE: [Asterisk-Users] no results.
the database schema is the table and it's associated columns. did you use
the create table script in the header of the pl file? basically, for each
of your sip entries, they would be broken down per line. so if your
sip.conf entry looks like this:
[1234]
type=friend
username=1234
secret=blah
nat=yes
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.0.4
your entries in the mysql database would be like this:
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'type', 'friend', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'username', '1234', '0');
INSERT INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to disable an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have
something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into
the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
i have been working with the retrieve_sip_conf_from_mysql.pl file and i
have
set everything as required. but when i run this script i am continuously
getting the no results in my screen and the file written by this
script
has only first result although i have many in my database. this is the
part
of this script.
my @resSet = @{$result};
print $#resSet;
if ( $#resSet == -1 ) {
print no results\n;
exit;
}
can any one tell me what is happening? and get rid of this error?
for those who have no clue.. this file is in the /usr/src/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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Re: [Asterisk-Users] Call recording
xlite saying login timed out. contact network admin. how to get rid of this. * is not behind NAT. DIAX works fine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP client not registering to *
xlite saying login timed out. contact network admin. how to get rid of this. * is not behind NAT. also, the grandstream SIP phone also seems to fail to register. IAX phones are all ok. DIAX works fine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call recording/SIP not loggin IN
My sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference dtmfmode=rfc2833 [xlite1] type=user host=dynamic secret=xlite1 context=outgoing reinvite=no canreinvite=no qualify=60 [xlite1] type=peer host=dynamic secret=xlite1 reinvite=no canreinvite=no qualify=60 In xlite i have User=xlite1, Pwd=xlite1 and SIP Proxy=IP of my * box, Out bound Proxy= IP of my * box netstat -na gives [EMAIL PROTECTED] root]# netstat -na Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 0.0.0.0:32768 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:22305 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:22273 0.0.0.0:* LISTEN tcp0 0 127.0.0.1:32769 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:33060.0.0.0:* LISTEN tcp0 0 0.0.0.0:111 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN tcp0 0 0.0.0.0:56800.0.0.0:* LISTEN tcp0 0 0.0.0.0:80 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:22321 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:22289 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:21 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:22 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:23 0.0.0.0:* LISTEN tcp0 0 0.0.0.0:443 0.0.0.0:* LISTEN tcp0128 202.51.xx.xx1:22202.51.xx.xx0:3148 ESTABLISHED udp0 0 0.0.0.0:32769 0.0.0.0:* udp0 0 0.0.0.0:50360.0.0.0:* udp0 0 0.0.0.0:50600.0.0.0:* udp0 0 0.0.0.0:45690.0.0.0:* udp0 0 0.0.0.0:111 0.0.0.0:* udp0 0 0.0.0.0:11770 0.0.0.0:* udp0 0 0.0.0.0:11771 0.0.0.0:* udp0 0 0.0.0.0:24270.0.0.0:* Active UNIX domain sockets (servers and established) Proto RefCnt Flags Type State I-Node Path unix 2 [ ACC ] STREAM LISTENING 1504 /dev/gpmctl unix 2 [ ACC ] STREAM LISTENING 1775 /tmp/.font-unix/fs7100 unix 2 [ ACC ] STREAM LISTENING 1520 /var/lib/mysql/mysql.sock unix 2 [ ACC ] STREAM LISTENING 1885 /var/run/asterisk.ctl unix 2 [ ACC ] STREAM LISTENING 1621 /tmp/.iroha_unix/IROHA unix 2 [ ACC ] STREAM LISTENING 1593 /tmp/cd_sockV4 unix 2 [ ACC ] STREAM LISTENING 1671 /tmp/kd_sockV4 unix 2 [ ACC ] STREAM LISTENING 1699 /tmp/td_sockV4 unix 2 [ ACC ] STREAM LISTENING 1565 /tmp/jd_sockV4 unix 7 [ ] DGRAM1094 /dev/log unix 3 [ ] STREAM CONNECTED 1889 /var/lib/mysql/mysql.sock unix 3 [ ] STREAM CONNECTED 1888 unix 2 [ ] DGRAM1778 unix 2 [ ] DGRAM1645 unix 2 [ ] DGRAM1406 unix 2 [ ] DGRAM1160 unix 2 [ ] DGRAM1110 [EMAIL PROTECTED] root]# my grandstream is also not registering to *. - Original Message - From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 02, 2004 9:14 PM Subject: Re: [Asterisk-Users] Call recording - Original Message - From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 02, 2004 9:30 AM Subject: Re: [Asterisk-Users] Call recording xlite saying login timed out. contact network admin. how to get rid of this. * is not behind NAT. DIAX works fine Could you especify a bit more? Send sip.conf, 'netstat -na' from you linux box, xlite config, etc... Regards, Gus ___ 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] SIP/grandstream not registering
hi, i can't seem to register my grandstream SIP to * server... i have my grandstream IP as 192.168.0.11 want to register to * at 202.51.xx.xxx. sip show peers says that my grand stream has unspecified IP but when i try to dial a number it gets this error... WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 40939 (Response) my sip.conf is... [grandstream2] type=peer host=dynamic secret=grandstream2 reinvite=no canreinvite=no qualify=60 [grandstream2] type=user host=dynamic secret=grandstream2 context=outgoing reinvite=no canreinvite=no qualify=60 help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SQL Updater Down!!!
hi, I am trying to install ASTGUICLIENT and when i run the AST_WINphoneAPP_0.8.pl it opens a window VICI Phone App -0.8 but i am getting SQL Updater Down Mesasge. How can i solve this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast gui client error
### connect to asterisk manager through telnet $t = new Net::Telnet (Port = 5038, Prompt = '/.*[\$%#] $/', Output_record_separator = '',); #$fh = $t-dump_log(./telnet_log.txt); # uncomment for telnet log $t-open($server_ip); i got error in this line $t-open($server_ip); my ip is 192.168.0.5 for asterisk and everyhings ok. the error i get is [EMAIL PROTECTED] astguiclient]# perl AST_SQL_update_channels.pl problem connecting to 192.168.0.5, port 5038: Connection refused at AST_SQL_update_channels.pl line 73 anyhting to do with port?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Config thru web interface or any GUI
hi, i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers? Suggest. Chandra
Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI
is there a installation guide? i didn't find any. just the readme file. - Original Message - From: Peter Brown To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 2:58 PM Subject: Re: [Asterisk-Users] Asterisk Config thru web interface or any GUI Chandra,Take a look at: http://sourceforge.net/projects/astguiclient/ it may be what your looking for or you could use the ideas if you want to make changes.I believe it was written by Matt Florell, Thanks Matt.At 14:42 30/12/03 +0545, you wrote: hi,i have been looking for any GUI that would make things easier to configure friends and peers into asterisk. I also looked at some posts in the lists. There are discussions that say text or CLI is more appropriate for adding users and stuff. Anyone know of any interface that would make things easier. How has NuFone or Voicepulse or IaxTel guys have implemented their asterisk box to add friends or peers?Suggest.Chandra Peter Brown
[Asterisk-Users] Grandstream budgetTone registration time out
hi, i have been using grandstream budgettone IP phones and they work fine except that these phones times out after some hours.. i ahve seen that the phones working ok are next day unregistered and sip show peers do not show their IP and although these phones can make calls , they cannot be called. They Sip show peers only shows their IP when i restart the IP phones. This is really annoying me now. Is there any better solutions than just restarting the phones every day? Any help is appreciated. cm
Re: [Asterisk-Users] SIP / X-ten Softphone
if u are using NAT, try adding nat=yes canreinvite=no qualify=500 in your [1005] - Original Message - From: Sean Cheesman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 19, 2003 9:31 AM Subject: RE: [Asterisk-Users] SIP / X-ten Softphone try adding username=1005 under [1005] and see if that helps -Original Message- From: PBX [mailto:[EMAIL PROTECTED] Sent: Thursday, December 18, 2003 10:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP / X-ten Softphone I know this has been covered more times than to mention and this is where I got most of my info from... But I am having issues with this. I can't seem to get the phone to register with *. This is being tested on a internal network right now. Here is the setup - sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow = all [1005] type=friend secret=1005 host=dynamic mailbox=1005 context=default extension.conf ; SIP exten = 1005,1,Dial,SIP/1005|15 exten = 1005,2,Voicemail2(u1005) exten = 1005,102,Voicemail2(b8200) X-Lite Softphone Network-Out Bound SIP Proxy: (IP of *) SIP Proxy-Default-Enabled: Yes SIP Proxy-Default-UserName: 1005 SIP Proxy-Default-Password: 1005 SIP Proxy-Default-Domain/Realm: (IP of *) SIP Proxy-Default-SIP Proxy: (IP of *) SIP Proxy-Default-Send Internal IP: Always But the only thing I every get is discovering firewall and Discovered Full Cone Nat Firewall. Any ideas... -gcc ___ 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] Re: * with RADIUS
- Original Message - From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 10:37 PM Subject: * with RADIUS hi, i have been looking for implementations of asterisk with RADIUS which would ease for accounting purposes. where can i find more information on this? help. cm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] restricting one user per account
last time i was experimenting IAXClient as a true client from dial up and registering it with * behind NAT. i found that the same account can b used by multiple ppl at the same time. is there any way to restrict one user per account i/e if on user has already exist then the another user with same account should be rejected? cm
