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[Asterisk-Users] Video Conference
Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough.
Re: [Asterisk-Users] Using * for large VoIP implementation
I have it on my network but it is still on a day-to-day evaluation case for me. Sometimes the quality is not at its best, probably because I am only using 800MHz CPU running SIP, Digium, T1-PRI, with g729 licenses. Maybe if you could spare time to put it on a high-end machine and evaluate, you'll find better results. Just take note that echo-cancellation, etc. of * is based on software (and is CPU intensive). Unlike existing VoIP gateways on the market, they have hardware chips (DSPs). Plus, most users of * is g711. Finding support for G729 is difficult though. For calling card application, it will be quite successful, as number of users for calling card is not as many as for wholesale. - Original Message - From: John Matte [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 21, 2004 11:32 AM Subject: [Asterisk-Users] Using * for large VoIP implementation Hi, I've searched the archives and can't find a satisfactory answer to this question: Does anybody out there have experience using * in large VoIP deployments? That is, switching large number of minutes from PSTN to VoIP and vice versa? How stable is it, what are the user experiences. I know Voicepulse and a few others do that, but I'd be interested to hear the experiences, what hardware is in use (e.g. who is using digium T1/E1 cards). Would greatly appreciate input on this Thanks _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ 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] Echo
I'm having bad echo between TDM and SIP.There's no echo between TDM-TDM though. I've seen this post from JTodd: ; Config notes:; - in /usr/src/zaptel/Makefile, set KFLAGS+=-DECHO_CAN_MARK2; - in /usr/src/zaptel/Makefile, set KFLAGS+=-DAGGRESSIVE_SUPPRESSOR ;; I compile with these two echo cancellation flags as it seems they; sound better with SIP phones interacting with Zap (analog) devices. I am not into compiling, so I need to verify if I will just edit the Makefile in /usr/src/zaptel folder and execute "makefw"? or just load ztcfg after modifying the Makefile. Thanks for the usual promptness. :) I welcome other ideas to resolve this echo issue.
[Asterisk-Users] G729 license
Hello all, I would like to just verify where to purchase the G729 license for Asterisk. Like I want to run G729 codec for all my calls passing thru Asterisk (voicemail, parking, via ZAP, via SIP, etc). The list says license is taken from Digium, does that apply also if I have Dialogic cards on my *?
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
Go for inter-fone products. it can both support sip and h323. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, January 23, 2004 2:40 PM Subject: [Asterisk-Users] Mediatrix 1204 sip experience? Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ 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] simple question...
it just came to my mind, and i haven't done any researches yet if somebody tried this one with asterisk.. :) well just in case somebody or someone on the list aware, i appreciate any advise. in telco world, there's like 64kbps per channel and voice can be carried on a 16kbps channel. is it possible to configure asterisk to make 4 extensions (ATAs example), to call out using single FXO port at the same time? if that is possible, then is it also possible to make t1-pri to be capable of transmitting 4x23ch simultaneous calls..? just curious.. nothing very serious. thanks :)
Re: [Asterisk-Users] Background Noise
yes, power is also source of background noise, but not all the time. you must also have proper grounding. with pc-phone, it could also be related to your phone/speaker gain controls. so u have to be definite on what kind of background noise u are getting. best codec for sip? it's always best to have alaw or ulaw. :) but bw-consuming. alaw is if u are using E1s. ulaw for T1s. don't even bother to use gsm. better get g729. some says u can use g723.1 with * w/o add'l purchase of license or something.. but i couldn't make this codec work. (so i went for g729.. :) ) im testing its quality though as digium cards are using CPUs as DSPs. - Original Message - From: Jonathan Biggs [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 11:01 AM Subject: Re: [Asterisk-Users] Background Noise Just to add some info from recent experience. May help, May not 1 X100P 2 X 4 port TDM400P Had to hook a dial-in palm PDA base for a custom software implementation to my * system and have the modem dial out and work properly. Phone connects to second port on PDA Base Experienced very bad electrical type noise on the line, hum, buzz, fad in and out. would come and go. Switched ports, wires, rxgain and txgain changes, phone changes, nothing helped the PDA base is also connected via serial port (or USB did not matter) to desktop computer for sync purposes Through trial and error. Found noise coming from connection to desk top computer. On, Off did not matter. Resolution. The power strip surge protector I was using on the desktop computer has two modes for noise filtration built in. 75 Hertz (or something, don;t remember) and 50 Hertz. I moved the plug for the desktop from the 75 side to the 50 side, All noise on line now gone. Not sure if this helps, constant noise on all sides may be power and noise filtration related... --- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi All i have some background noise problem with * and a diva srv 4bri + chan_capi 0.3.0 + X-Ten PRO on my pc. Both in incoming and outgoing call have a background noise. there is some tuning to do? where can i find documentation about capi.conf? which is the best codec for sip (ulaw, alaw, gsm...)? mark ___ 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 __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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] SER Asterisk
I tried the following setups. Although there is a minor ringback issue that I haven't found any solution yet. 1.) ATA-CiscoNAT-Asterisk-SER+RTPProxy-Cisco2600 2.) ATA-CiscoNAT-SER+RTPProxy1-Asterisk-SER+RTPProxy2-Cisco2600 I cannot remember if * can directly connect to Cisco2600. I know I had problems initially with it, that's why I installed the SER, and since now I'm focusing to solve the ringback issue, I didn't have time to take out SER out of my equation. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 16, 2004 7:33 PM Subject: Re: [Asterisk-Users] SER Asterisk Thanks guys, thought SER had to 'register' to be able to use any Asterisk contexts. But just defining a new entry in the sip.conf with just context ip worked! But now i'm stumbling on another problem.. Asterisk seems to want to send the SIP udp packets directly to the SIP clients. In the case of a SIP user/client behind a NAT, this obviously doesn't work. SER is configured to use the wonderful RTPProxy + SER nathelper module, and this works flawlessly (using the rewritehostport function). But when I try to call a phone number on the PSTN network from a SIP client behind NAT, SER sends the invites to Asterisk, and Asterisk makes an outbound call to the phone number, the phone rings, but when the pstn user picks up the phone, no sound, and after a while (couple of seconds), the call is dropped. Asterisk spews out the following warning, chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 29898 (Response) Tried searching on the voip-info wiki and mailinglists, but didn't find a way to force Asterisk to use a SIP proxy/SER. Any ideas ? On Fri, Jan 16, 2004 at 12:12:14AM -0800, Chris Albertson wrote: Yes, you can keep non-authorized SIP callers from accessing the PSTN by setting up the .conf file correctly as below but you can also run a fire wall on the box that Asterisk runs on. Firewall off SIP ports except for if they come from your SER server. --- Fran Boon [EMAIL PROTECTED] wrote: [ser] context=sip-legal host=y.y.y.y ; IP address of SER Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO ___ 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] Ringback Problem
I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN debug from Cisco. It's sending ALERT (progress indicator 8), but I am not sure why * is not relaying that signal to the ATA. I hope someone is out there to help me on this. Thanks in advance. Jan 15 22:53:24: ISDN Se1/0:23: TX - SETUP pd = 8 callref = 0x0078Jan 15 22:53:24: Sending CompleteJan 15 22:53:24: Bearer Capability i = 0x8090A2Jan 15 22:53:24: Channel ID i = 0xA98397Jan 15 22:53:24: Calling Party Number i = 0x0081, '6882332', Plan:Unknown, Type:UnknownJan 15 22:53:24: Called Party Number i = 0x80, '12123637055', Plan:Unknown, Type:UnknownJan 15 22:53:24: ISDN Se1/0:23: RX - STATUS pd = 8 callref = 0x8078Jan 15 22:53:24: Cause i = 0x80E3A1 - Information element not implementedJan 15 22:53:24: Call State i = 0x01Jan 15 22:53:24: ISDN Se1/0:23: RX - CALL_PROC pd = 8 callref = 0x8078Jan 15 22:53:24: Channel ID i = 0xA98397Jan 15 22:53:27: ISDN Se1/0:23: RX - ALERTING pd = 8 callref = 0x8078Jan 15 22:53:27: Progress Ind i = 0x8088 - In-band info or appropriate now available Jan 15 22:53:32: ISDN Se1/0:23: TX - DISCONNECT pd = 8 callref = 0x0078Jan 15 22:53:32: Cause i = 0x8090 - Normal call clearingJan 15 22:53:32: ISDN Se1/0:23: RX - RELEASE pd = 8 callref = 0x8078
[Asterisk-Users] announcement using Dial
IF I want to play sound files, 1.) what format should it be? (*.au or*.wav) 2.) where should it reside? 3.) what syntax should I follow? Is exten=_.,102,Dial(SIP/[EMAIL PROTECTED],1,tHA(sound.au)) correct? I tried this and it doesn't work. Thanks,
Re: [Asterisk-Users] Specifying a codec to be used in /etc/sip.conf
Follow-up question, what does * use for fax? T38 or passthrough? - Original Message - From: Peter Bittner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:12 AM Subject: [Asterisk-Users] Specifying a codec to be used in /etc/sip.conf Hi all! Is it possible to tell * to allow connecting an incoming (SIP-) call with the G711 codec (a simple fax). I have not found any setting in sip.conf that would refer to this problem. I am using * and the spandsp library to receive faxes from a SIP gateway. Everything works for now except the final transmission of the fax. It seems that the sender and *, the receiver, do not negotiate the correct codec, which must definitely be G711. Any ideas? Peter ___ 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] * and telco ringback
Anybody knows why I am not receiving any telco alerting (ringback, fast busy, etc)? I can receive the rtp for voice though. Pls advise thank you.
Re: [Asterisk-Users] How does the PSTN termination indurty realy work?
normally, voip companies have their own switches, meaning PoP located locally to resell minutes to other carriers, or they may have what they call partners residing anywhere in the globe where they buy minutes from to push their traffic to. in order for an individual to operate fully as voip operator, you will need lots of back-up routes, bec most of voip operations are not legal outside US -- and there are ppl not afraid of closing down after 2-6 months because they already made money out of their routes during the operation. anyway, to put up your own company u need lots of contacts, cold calls, (gateways if you want), and the most important thing is softswitch+billing. with softswitch u can buy/resell minutes between voip vendors/clients. and of course billing is for your own sake, otherwise, you'll loose everything. :) good luck, and let me know if you got good routes/rates. right now im looking for cuba for additional capacity. - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 3:13 PM Subject: [Asterisk-Users] How does the PSTN termination indurty realy work? Does anyone here understand how these VOIP/PSTN gateway companies work? There seem to be a number of small outfits that offer about the same service. I'm thinking that a company that is really just one person and a web site can't possibly own and operate gateway equipent in 100 contries world wide. I figure there must be some large outfit who _does_ have physical equipment installed in hundreds of locations that leases time to re-sellers like Nufone, Iconect, Voice Pulse, Addaline, xvoip and the others If this is realy how the industry works, then does it matter which retailer you pick? Matter in terms of audio quality and reliabilty, this is. I know each will have it's own customer support and pricing. This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I am having probelms connecting to voicepulse this morning. Is anybody else = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ 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] * and signaling (clarification)
Hello to the list again.
I have my ATA behind NAT connecting to * then calls
are fwd to Cisco 2600. Calls are completing, I just cannot figure out why
I don't hear any ALERTING signals from the 2600 (ringback, fast busy, SIT,
etc). Audio works fine though. I'm using G711ulaw. And I don't
want ATA or * to provide the false ringback so I took out ('r') in my Dial
command. Existing command is as follows:
exten=_.,1,Dial(SIP/${EXTEN},20,tH)
I hope somebody out there could help me on
this. Thanks.
[Asterisk-Users] SIP/2.0 487 Request Cancelled
Here's my sip debug output. anybody knows whyCisco sent * isCANCEL msg? Can someone tell me what ATA version are they using? Maybe this is also another issue.. I am using v2.16. This is using G711ulaw. Sip read: SIP/2.0 100 TryingVia: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3eFrom: "Jess" sip:[EMAIL PROTECTED];tag=as6818ebfbTo: sip:[EMAIL PROTECTED];tag=243881BC-1D1Date: Fri, 09 Jan 2004 17:56:09 GMTCall-ID: [EMAIL PROTECTED]Server: Cisco-SIPGateway/IOS-12.xCSeq: 102 INVITEAllow-Events: telephone-eventContent-Length: 0 10 headers, 0 linesSip read: SIP/2.0 200 OKVia: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3eFrom: "Jess" sip:[EMAIL PROTECTED];tag=as6818ebfbTo: sip:[EMAIL PROTECTED]Date: Fri, 09 Jan 2004 17:56:09 GMTCall-ID: [EMAIL PROTECTED]Content-Length: 0CSeq: 102 CANCEL 8 headers, 0 linesSip read: SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3eFrom: "Jess" sip:[EMAIL PROTECTED];tag=as6818ebfbTo: sip:[EMAIL PROTECTED];tag=243881BC-1D1Date: Fri, 09 Jan 2004 17:56:09 GMTCall-ID: [EMAIL PROTECTED]Server: Cisco-SIPGateway/IOS-12.xCSeq: 102 INVITEAllow-Events: telephone-eventContent-Length: 0 - FROM CISCO - Jan 9 17:20:39: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2 From: "Jess" sip:[EMAIL PROTECTED];tag=as7f1f229e To: sip:[EMAIL PROTECTED];tag=241801E4-2525 Date: Fri, 09 Jan 2004 17:20:39 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 Jan 9 17:20:39: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDINGJan 9 17:20:39: ccsip_report_digit_control: enable=0: Jan 9 17:20:39: ccsip_report_digit_control: disabled.Jan 9 17:20:39: *CCB found in UAS Request table. ccb=0x642D4630Jan 9 17:20:39: CCSIP-SPI-CONTROL: act_recdinvite_new_messageJan 9 17:20:39: CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT: Using GMTJan 9 17:20:39: sip_stats_methodJan 9 17:20:39: ccsip_set_release_source_for_peer:ownCallId[1132803], src[2] Jan 9 17:20:39: 0x642D4630 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)Jan 9 17:20:39: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGEJan 9 17:20:39: sip_stats_status_codeJan 9 17:20:39: CCSIP-SPI-CONTROL: sipSPISendInviteResponseJan 9 17:20:39: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGEJan 9 17:20:39: sip_stats_status_codeJan 9 17:20:39: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for incoming callJan 9 17:20:39: 0x642D4630 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)Jan 9 17:20:39: Ignoring unexpected event 11 (SIPSPI_EV_CC_CALL_PROCEEDING) in state 8 (STATE_DISCONNECTING) substate 0 (SUBSTATE_NONE)Jan 9 17:20:39: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECTJan 9 17:20:39: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2 From: "Jess" sip:[EMAIL PROTECTED];tag=as7f1f229e To: sip:[EMAIL PROTECTED] Date: Fri, 09 Jan 2004 17:20:39 GMT Call-ID: [EMAIL PROTECTED] Content-Length: 0 CSeq: 102 CANCEL Jan 9 17:20:39: Sent: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2 From: "Jess" sip:[EMAIL PROTECTED];tag=as7f1f229e To: sip:[EMAIL PROTECTED];tag=241801E4-2525 Date: Fri, 09 Jan 2004 17:20:39 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0
Re: [Asterisk-Users] * and Cisco Gateways
Testing between ATA and Asterisk is working fine. I am getting voicemail etc. But when I'm trying to call to the carrier side i find it not working. I see on my Cisco gateway that it negotiated the g711ulaw codec, but when the state goes into active, I just automatically get busy tone from my ATA. It looks like that the ATA-* leg is disconnected while *-CiscoGateway is still trying to connect. My ATA is setup as g711ulaw (using Txcodec:2, Rxcodec:2). Probably * is trying to negotiate with ATA using GSM codec? Is there a command in * (or * utility) that I can use to debug the codec negotiation, and/or the RTP status? My sip and exten are as follows: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = siptest tos=lowdelay tos=184 maxexpirey=3600 disallow=all allow=gsm allow=ulaw register = [EMAIL PROTECTED] nat=yes query=yes autocreatepeer=yes [carrier] type=friend fromdomain=mydomain host=X.X.X.X - carrier IP extensions.conf ;extension exten = 1234,1,Dial(SIP/1234,10,tr) exten = 1234,2,Voicemail,u1234 exten = 1234,102,Voicemail,b1234 ;oubound to carrier exten = _011.,1,Dial(SIP/[EMAIL PROTECTED],tr) ;getting voicmails exten = 73*,1,VoicemailMain - Original Message - From: Arslan Saeed [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 6:55 PM Subject: RE: [Asterisk-Users] * and Cisco Gateways Hi, Send me your * configuration u r using for troubleshooting this problem. Arslan. -Original Message- From: Jess Magnaye [mailto:[EMAIL PROTECTED] Sent: Friday, January 09, 2004 1:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * and Cisco Gateways Importance: High That didn't work. I ran tcpdump and looks like looks like my Linux has problems with its on-board Ethernet. I'm getting the following error, which most likely is the reason for no audio. 14:40:57.628080 asterisk cisco: icmp: asterisk udp port 13696 unreachable [tos 0xd4] 14:40:58.275665 asterisk cisco: icmp: asterisk udp port 13697 unreachable [tos 0xc0] 14:40:58.622308 asterisk cisco: icmp: asterisk udp port 13696 unreachable [tos 0xd4] I am now installing a new linux with (pci-ethernet) to load the *. I hope when I'm done, things will go smoothly. - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 1:32 PM Subject: Re: [Asterisk-Users] * and Cisco Gateways Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following codec negotiation problem from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks. Go into sip.conf, and add these lines to the SIP peer for your Cisco 2600: disallow=all allow=ulaw allow=alaw This will force G.711 codec usage, which may solve your problems though it will increase your bandwidth. JT ___ 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 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 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] At last!!! :)
I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config was: exten=_.,1,Dial(SIP/[EMAIL PROTECTED],tr). The reason why my ATA is getting fast busy (or dropping the call immediately) while Cisco gateway (myprovider) is trying to connect my call, was that I am missing the "seconds" parameter. When I changed this to Dial(SIP/[EMAIL PROTECTED],20,tr), I was able to connect. There is one little problem left though. How come after I diale the number from ATA, I am getting false ringback. I meant, local ringback from ATA, instead of the ringback coming from my Cisco (myprovider). I appreciate any bright ideas and advise from anybody. Thank you and have a happy weekend!
Re: [Asterisk-Users] PSTN SIP Gateways?
i worked with Mediatrix boxes before (in '01-'02). configuring them using the ume (unit manager express) is highly suggested, as there are parameters in there that you won't see in its telnet access... :) generally it works fine, depending on how you use them, or how strict are you in evaluating a product. i used it for both origination (fxs) and termination (fxo). for termination, they only supported battery-reversal that time. i am not aware now if they support answer supervision via voice-detect. a particular problem i remember with the termination is that it cannot handle properly the far-end disconnect, where it locks up or the channel appears to be in a call even if the far-end already got disconnected. for fxs, i had problems with the volume (rx/tx gain) -- they're inconsistent -- something like, you'll find them working at its best today, and the next day you don't know what happens why the quality is not the same as it was. (so maybe they made some improvements at the present? u have to find out. :) good luck. ) - Original Message - From: Michael Graves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 09, 2004 5:56 PM Subject: [Asterisk-Users] PSTN SIP Gateways? Since my earlier inquiry about gateways went unanswered perhaps rephrasing will help. Does anyone here have experience with standalone SIP FXO gateways like those from Mediatrix? Care to share their experiences with them? Off list if necessary. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] I knew a man who couldn't lose 'cause he never gave in. He stuck to his pistols and it made him a better man. - D. Van Zant ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] E100P : Pb with outgoing calls
check why is this happening: Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] it looks to me that your operator doesn't recognize a message coming from the *. sometimes this is fixed by changing the switch-type. you may need to get your operator involve on this. i experienced this error too when i interconnected my *'s t1-pri to cisco-pri, and it was resolved when i changed my cisco's config to use bearer-cap speech. i am not sure if that will help in your case, or if you will need to verify with * or with your operator about the bearer capability issues. try also checking the isdn plan/type. i see it here that it is using national/isdn. try setting it up as unknown/unknown. - Original Message - From: Olivier Perrin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 4:12 AM Subject: [Asterisk-Users] E100P : Pb with outgoing calls I use a E100P in France with a french operator E1. I can receive calls via the E1 and tranfer them to a VoIP phone, play IVR etc But outgoing calls doesn't work at all. I receive a RELEASE COMPLETE just after the SETUP. There is no pb with the operator (the E1 work well with an other Pbx). Here a call trace. Anyone have an idea ? (g1 is my group name for the 30 channels) -- Accepting call from '147241527' to '5797' on channel 21, span 1 -- Executing Dial(Zap/21-1, Zap/g1/3361100) in new stack -- Making new call for cr 32784 Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '147241527' ] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3361100' ] Sending Complete (len= 0) -- Called g1/3361100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32784/0x8010) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] -- Processing IE 8 (Cause) -- Channel 1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == No one is available to answer at this time ___ 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] * and Cisco Gateways
Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.
Re: [Asterisk-Users] * and Cisco Gateways
That didn't work. I ran tcpdump and looks like looks like my Linux has problems with its on-board Ethernet. I'm getting the following error, which most likely is the reason for no audio. 14:40:57.628080 asterisk cisco: icmp: asterisk udp port 13696 unreachable [tos 0xd4] 14:40:58.275665 asterisk cisco: icmp: asterisk udp port 13697 unreachable [tos 0xc0] 14:40:58.622308 asterisk cisco: icmp: asterisk udp port 13696 unreachable [tos 0xd4] I am now installing a new linux with (pci-ethernet) to load the *. I hope when I'm done, things will go smoothly. - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 1:32 PM Subject: Re: [Asterisk-Users] * and Cisco Gateways Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following codec negotiation problem from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks. Go into sip.conf, and add these lines to the SIP peer for your Cisco 2600: disallow=all allow=ulaw allow=alaw This will force G.711 codec usage, which may solve your problems though it will increase your bandwidth. JT ___ 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2nd call leg status?
of course you must use a digital outbound interface (bri,pri), since on analog there's no way to know if the call has been answered (and for cdr logs asterisk assumes that the call in always ANSWERED when dialling through analog lines). does that mean battery reversal is not supported by * for analog interfaces?? ___ 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] * crashed
I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this error when im trying to connect, and then it suddenly dies out. ERROR[1074412224]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk I tried to run it again using "asterisk -gc" and I got the ff error: WARNING[1074412224]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1Segmentation fault (core dumped) Not sure why. It looks to me it got corrupted after my reboot during change of IP. (Can someone shed light on this?) Thanks.
Re: [Asterisk-Users] * crashed
hmm.. i did the modprobe wct1xxp. but i didn't do ztcfg bec i thought i will only need it if there's change in the zaptel config. let me try it and i'll let u know. thanks. - Original Message - From: Brent Franks To: [EMAIL PROTECTED] Sent: Wednesday, January 07, 2004 11:52 AM Subject: RE: [Asterisk-Users] * crashed Are you doing these items before trying to start asterisk: loadmod driver (In my case its wcfxo) After that is completed: Ztcfg This configures all of the zaptel harware in the system. You might want to place this in init.d to get it to do it automatically. - Brent -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jess MagnayeSent: Wednesday, January 07, 2004 11:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] * crashedImportance: High I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this error when im trying to connect, and then it suddenly dies out. ERROR[1074412224]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk I tried to run it again using "asterisk -gc" and I got the ff error: WARNING[1074412224]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1Segmentation fault (core dumped) Not sure why. It looks to me it got corrupted after my reboot during change of IP. (Can someone shed light on this?) Thanks.
Re: [Asterisk-Users] Pls confirm
Does this mean I can run with G711 between ATA and *, and GSM between * and voip-provider? Example: ATA - g711 via SIP - * - gsmfr via SIP - Cisco-VoIP - Pots I am wondering because I am just getting silence. It looks like rtp coming back from the voip-provider is not matching with my * or ATA's codec. - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 2:50 PM Subject: Re: [Asterisk-Users] Pls confirm On Tuesday 06 January 2004 12:00, Jess Magnaye wrote: Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? G.723 and G.729 are both patent-encumbered codecs. G.729 is available for purchase by contacting Digium. G.723 is not. I would really recommend that you go with a different codec, such as GSM or ILBC. -Tilghman ___ 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] * crashed
Using modprobe+ztcfg didn't work. Kernel patch is ok. I just re-installed aterisk and now it's working. :( Anyway, now that my * is remotely placed. My ATAs are behind NAT. (setup is: private ATA - router+nat - public internet - public*). I configured my sip.conf to have host=172.30.200.27, nat=yes and qualify=yes. Unfortunately, I'm still getting this message (no NAT) on my debug. 9 headers, 0 lines Using latest request as basis request Sending to 172.30.200.27 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.200.27:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=523636298 To: sip:[EMAIL PROTECTED];user=phone;tag=as301ad6f2 Call-ID: [EMAIL PROTECTED] CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 - Original Message - From: Michael Devenijn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 07, 2004 11:05 AM Subject: RE: [Asterisk-Users] * crashed May be it's due to a kernel patch ... ?? Try to recompile zaptel, asterisk, ... Van: Jess Magnaye [mailto:[EMAIL PROTECTED] Verzonden: wo 7/01/2004 17:36 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] * crashed I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this error when im trying to connect, and then it suddenly dies out. ERROR[1074412224]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk I tried to run it again using asterisk -gc and I got the ff error: WARNING[1074412224]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Segmentation fault (core dumped) Not sure why. It looks to me it got corrupted after my reboot during change of IP. (Can someone shed light on this?) Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pls confirm
Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711?
Fw: [Asterisk-Users] Pls confirm
- Original Message - From: Jess Magnaye [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm Is the format allow=g723.1 in sip.conf valid? somehow i cannot get it working to do g723 passthru. also, i've read that doing g723 will disable voicemail, and other * features. is that true? if it is, will the licensed-g729 work for * features? thanks again in advance. - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 1:38 PM Subject: Re: [Asterisk-Users] Pls confirm Jess Magnaye wrote: Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? It can do G.723 between endpoints (passthrough).. It can do G.729a with the purchase of an additional licence of $10 per channel.. Yes you can use G.711 with a provider, some providers offer GSM, iLBC and Speex as alternative codecs.. Later.. ___ 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] question re voicemail
Hi,
I just setup my * with digium. I started testing
voicemail first between atas, and i am not sure why it is not prompting me any
when the call is not answered or if busy.i only get continuous
ringback andthe following message:
asterisk*CLI --
Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new
stack -- Called 5104112978 --
SIP/5104112978-3f88 is ringing -- Nobody picked up in
2 ms
I wonder if my uextension and
bextension config is correct, mispelled, or something else is
missing. Note that ata to ata via * works,
as well as getting to VoicemailMain via extension 1234. Please
help. My config are found below. I appreciate your
help.
sip.conf
---
[6882332]type=friendusername=6882332secret=testhost=dynamicdefaultip=172.30.200.27dtmfmode=rfc2833mailbox=6882332callerid
= "test1" 6882332context=sip
[5104112978]type=friendusername=5104112978secret=testhost=dynamic;canreinvite=nodefaultip=172.30.200.26dtmfmode=rfc2833mailbox=5104112978callerid
= "test2" 5104112978context=sip
extensions.conf
[sip];ringexten =
5104112978,1,Dial(SIP/5104112978,20,tr)exten =
6882332,1,Dial(SIP/6882332,15,tr)exten =
,1,Dial(SIP/,5,tr)
;unansweredexten =
6882332,102,Voicemail,u6882332exten =
5104112978,102,Voicemail,u5104112978exten =
,102,Voicemail,u
;busyexten =
6882332,103,Voicemail,b6882332exten =
5104112978,103,Voicemail,b5104112978exten =
,103,Voicemail,b
;get messageexten =
1234,1,VoicemailMain(6882332);exten =
,1,VoicemailMain();
voicemail.conf
-
[default]6882332 =
6882332,test1,[EMAIL PROTECTED]
5104112978 = 5104112978,test2, [EMAIL PROTECTED]
9011 = 9011,Asterisk,[EMAIL PROTECTED]
= ,Nada,[EMAIL PROTECTED]
[Asterisk-Users] Need Help...
Is this setup possible? ATA - Asterisk - SER with RTPproxy - AnySIPGateway how can I instruct * to send all unknown extensions to go to SER/RTPProxy? Do I have to use "exten"? What syntax shld I use? I cannot find any matching document from wikki. :(
Re: [Asterisk-Users] Are messages censored on this board?
No it is not censored. I think there is delay. - Original Message - From: John Coll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 05, 2004 5:23 PM Subject: [Asterisk-Users] Are messages censored on this board? I've submitted a message twice this evening and it has not appeared. Are messages censored on this board? regards john - John A Coll, Director, Connection Software 391 City Road, LONDON, EC1V 1NE, UK Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000 Fax: 020 7713 8001 Fax: +44 20 7713 8001 Email: [EMAIL PROTECTED] Web: www.csoft.co.uk PGP Public Key from keyserver ___ 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
Fw: [Asterisk-Users] Fw: Questions and finding
Thanks for the reply. 1. My VAD is turned off (00140014), and it didn't help for that cut-off. I am not sure if OutboundProxy has to be configured to have it working fine. Or this just happened to me? What is your ATA's software? 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. As per ATA, it is by default using rfc2833. I tried setting it up as inband by setting Audiomode, but nothing helped. I was thinking the * is ONLY recognizing the DTMF if there is telco board installed. Is it? - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: Jess Magnaye [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 12:36 PM Subject: Re: [Asterisk-Users] Fw: Questions and finding Hi! 1.) First test - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off after 5-10seconds (consistently). - Solution: I have to reconfigure ATA to use OutboundProxy to be Asterisk IP. - Am I doing the right thing? Turn of silence detection / VAD. Any solution to this one? My thinking was that DTMF can only be detected by * Take a look at your SIP configuration and make sure you have the correct dtmfmode= set. Try different values if you continue to have trouble and configure your ATA accordingly. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Questions and finding
Hi! 1. My VAD is turned off (00140014), and it didn't help for that cut-off. Then check if you have a firewall in between * and your ATA that closes the port due to inactivity of your ATA. Also use SIP DEBUG in the CLI to try to see a bit more of what is going on. You could also use Ethereal to monitor the SIP traffic (or the rtp/UDP traffic). am not sure if OutboundProxy has to be configured to have it working fine. Or this just happened to me? What is your ATA's software? I don't have such a device, in fact never had. :-) MY ATA and * are sitting on the same LAN. So FW or NAT problem is not possible. This is also the reason why I commented out nat=1 in the sip.conf. 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. Note: inband only works with g.711 codec. Doesn't the ATA also offer info as third dtmfmode option? Anyway, you might want to search the mailing list for setup info, there are a lot of people around that use it. by setting Audiomode, but nothing helped. I was thinking the * is ONLY recognizing the DTMF if there is telco board installed. Is it? No no, * doesn't require any hardware to be installed. LET ME TRY dtmfmode=info AND SEE WHAT HAPPENS NEXT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Questions and finding
Hi Philip, I found the problem. My sip.conf config was changed by somebody else. :( The external IP was uncommented and that's what is causing my problem. - Original Message - From: Jess Magnaye [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, December 23, 2003 4:33 PM Subject: Re: [Asterisk-Users] Fw: Questions and finding Hi! 1. My VAD is turned off (00140014), and it didn't help for that cut-off. Then check if you have a firewall in between * and your ATA that closes the port due to inactivity of your ATA. Also use SIP DEBUG in the CLI to try to see a bit more of what is going on. You could also use Ethereal to monitor the SIP traffic (or the rtp/UDP traffic). am not sure if OutboundProxy has to be configured to have it working fine. Or this just happened to me? What is your ATA's software? I don't have such a device, in fact never had. :-) MY ATA and * are sitting on the same LAN. So FW or NAT problem is not possible. This is also the reason why I commented out nat=1 in the sip.conf. 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. Note: inband only works with g.711 codec. Doesn't the ATA also offer info as third dtmfmode option? Anyway, you might want to search the mailing list for setup info, there are a lot of people around that use it. by setting Audiomode, but nothing helped. I was thinking the * is ONLY recognizing the DTMF if there is telco board installed. Is it? No no, * doesn't require any hardware to be installed. LET ME TRY dtmfmode=info AND SEE WHAT HAPPENS NEXT. ___ 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] Asterisk as a PSTN gateway for SER
how did u setup your asterisk for this: I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 2:42 PM Subject: [Asterisk-Users] Asterisk as a PSTN gateway for SER First off, here is what I want to do: SIP Clients - SER - Asterisk - VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients can talk to each other. I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: Here is the output: -- Executing Dial(SIP/-08114560, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/SIPprovider-5e0c is making progress passing it to SIP/-08114560 -- SIP/SIPprovider-5e0c answered SIP/-08114560 -- Attempting native bridge of SIP/-08114560 and SIP/SIPprovider-5e0c I have tried this with my SIP client behind a NAT and outside of a NAT, so I don't that is the problem. I have also tried this with both IAX and SIP providers and the problem is the same. One ring, and then silence. Any thoughts? Thank you for your time. __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ 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] Fw: Questions and finding
I installed * to primarily test its voicemail
feature. I installed it on a server WITHOUT any telco board (i.e.,
digium). Installation looks ok, however I am having problems.
MY SETUP:
2xATAs are configured to use * as
GkorProxy
Asterisk is registered to my SER SIP/RTP
Proxy
1.) First test
- ATA1 calls to ATA2. When voicemail starts
playing, it just cuts-off after 5-10seconds (consistently).
- Solution: I have to reconfigure ATA to use
OutboundProxy to be Asterisk IP.
- Am I doing the right thing?
2.) Second test:
- ATA1 calls ATA2 and left a message. * sends
an email to the owner of ATA2.
- ATA2 retrieves the message by dialing the
extension (i.e., exten=,1,VoiceMailMain(123))
- VoiceMail prompts for password. ATA2 keys
in its pasword. * doesn't reply and keeps on prompting for 5 times, until
it disconnects:
-- Executing
VoiceMailMain("SIP/6882332-6b3a", "123") in new stack --
Playing 'vm-password' (language 'en') -- Incorrect
password '' for user '123' (context = any) --
Playing 'vm-incorrect' (language 'en') -- Playing
'vm-password' (language 'en') -- Incorrect password '' for
user '123' (context = any) -- Playing
'vm-incorrect' (language 'en') -- Playing 'vm-password'
(language 'en')
- Tried to change AudioMode of ATA, to use
0x00140014, 0x11241124, and default 0x00150015. But there were no
difference.
- Tried to skip the password on *, using
exten=,1,VoiceMailMain(s123). * prompted there arenew
messages, and instructions on how to retreive them. Unfortunately,
none of the dtmf keys worked (1, #, etc).
Any solution to this one?
My thinking was that DTMF can only be detected by *
3.) How can I configure * to support the following
setup. I know * doesn't work 100% like softswitch, but maybe this type of
connection is supported?
ATA - * - CiscoAS5300
ATA to *, is IP connection
* to CiscoAS5300 is also IP connection
I appreciate your help. Thank
you.
