Re: [Asterisk-Users] weird call transfer problem
Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ 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] attension mark spencer
hi , I was wondering if i can get some algo or architecture of asterisk...i mean how different channels are working (specially agents,h323)and how call is established... i know i am sounding a bit stupid but i need this ...can you please guide me thanx Amna Saleem ___ 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] VAD/DTX implementation through zaptel cards
Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. Regards, Steve ___ 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 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] Polycom V500 With Asterisk Setup
Dear All, We've got 1 set of Polycom V500 Video Conferencing Kit in my Office. I'm trying to link it with Asterisk and is facing some issues. Would like to seek your kind advise. The Polycom V500 is unable to make the outgoing calls, and will always report the ENTER ERROR HERE. sip show peers does not shows that the Polycom V500 being able to register. The account is working alright as I've used the account on Eyebeam and its working fine. Here are the debug logs for the System -- SIP read from 192.168.100.146:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Polycom V500 Release 7.5 - 15Dec2004 10:12 Contact: sip:192.168.100.146 Content-Type: application/sdp Content-Length: 899 v=0 o=Vigor11 1627471320 0 IN IP4 192.168.100.146 s=- c=IN IP4 192.168.100.146 b=AS:384 t=0 0 m=audio 49178 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:98 SIREN14/16000 a=fmtp:98 bitrate=32000 a=rtpmap:97 SIREN14/16000 a=fmtp:97 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 m=video 49180 RTP/AVP 109 34 96 31 b=TIAS:384000 a=rtpmap:109 H264/9 a=fmtp:109 profile-level-id=42800c max-mbps=1 a=rtpmap:34 H263/9 a=rtpmap:96 H263-1998/9 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T a=rtpmap:31 H261/9 a=fmtp:31 CIF=1 QCIF=1 m=data 49182 RTP/AVP 100 a=rtpmap:100 H224 --- (11 headers 35 lines)--- Using latest request as basis request Sending to 192.168.100.146 : 5060 (non-NAT) Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5947 check_user_full: Setting NAT on RTP to 524288 Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5951 check_user_full: Setting NAT on VRTP to 524288 Reliably Transmitting (NAT) to 192.168.100.146:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655;received=192.168.100.146;rport=5060 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED];tag=as36644353 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: nVoice PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=nvoice, nonce=60b31ab3 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '898' tannery*CLI -- SIP read from 192.168.100.146:5060: ACK sip:192.168.100.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:192.168.100.146 Content-Length: 0 Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Version 0.80 of IPS released
| On a slightly different note: | | Is there a setting to force IPS not to minimise every time an action | is performed? | | It gets very annoying after a few minuites and with our reception | being very very busy it could get quiet sickly | On the config page: uncheck minimize after call/transfer Thorben ___ 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] Problem with fxo
Hi Moises
thanks for the help
but i have the same problem
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
this is my extension for dialing out
still the same weird exception 15 error :/
and group 1 es the one on my zapata.conf
starting to think about hardware problem :/
El mar, 12-04-2005 a las 14:20 +, Moises Silva escribió:
I have no Idea of the strange errors, but as far as i know, the proper
way of calling is:
Zap/g${group}/${phone_number}
where ${group} is a valid group inside zapata.conf, and
${phone_number} is the desired PSTN phone to call. In you email you
wrote the messages and i can see that you missed the letter 'g'
before the group and the last '/' slash. Give that a try, may be will
work.
Best Regards
- Moy
On Apr 12, 2005 11:23 AM, Julio Saura [EMAIL PROTECTED] wrote:
Hi,
i am trying to use my fxo card for analog calls ..
fxo card seems to be ok, working properly but when trying to call
outside ( from a sip phone ot pstn ) i get the following error on
asterisk .
Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route:
Contact hop: Drugo sip:[EMAIL PROTECTED]:5060
-- Executing Dial(SIP/69-562c, Zap/1/651559526|5) in new stack
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1645 zt_call: Dialing
'651559526'
Apr 12 11:59:24 DEBUG[4231]: chan_zap.c:1706 zt_call: Deferring
dialing...
-- Called 1/651559526
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:25 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Hook Transition Complete(12) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3770 __zt_exception: Exception
on 15, channel 1
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:3082 zt_handle_event: Got event
Dial Complete(9) on channel 1 (index 0)
Apr 12 11:59:27 DEBUG[4231]: chan_zap.c:1224 zt_enable_ec: Enabled echo
cancellation on channel 1
Apr 12 11:59:27 DEBUG[4231]: channel.c:1363 ast_read: Dropping duplicate
answer!
any clue?
got no info about exception 15 :/
Thanks in advance
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Re: [Asterisk-Users] Attempting native bridge of
On Apr 12, 2005, at 9:38 PM, snacktime wrote: That would be great if I didn't want * to get out of the media path, but I do. In my case everything works great with the teliax 800 DID, but not with the local number DID. I think it's an issue on their end myself. ___ I didn't want to insinuate that Teliax was in any way sloppy, but they *are* the ITSP I was referring to when I mentioned earlier in this thread that my provider was having issues with native bridging. I raised a ticket with them and they're working on resolving the bug currently, so I think you're in the same boat with me here Mr. Snackie. When I get closure on the ticket I'll send you a note and mention it in this thread for future searchers. /rg ___ 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] New PRI install with new te110p
Getting this error on a new install, I am lost since this is my first time messing with the te110p and my first PRI install. I have signalling=pri_cpe as the Digium docs suggest, when I start Asterisk I get this over and over: == Primary D-Channel on span 1 down Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No D-channels available! Using Primary on channel anyway 24! If I change signalling to pri_net the errors go away, either way I can receive calls into Asterisk. How should the signalling be set, to cpe or net? Any idea what's causing this error? I am not entirely sure my PRI is 100% up even, * seems to be talking to it because when I pull the cable it starts giving me alerts and such, the alerts go away when I plug the cable back in. Of course the telco is waiting for me to call them so we can test the PRI against my equipment.. I guess they expect me to have known working equipment.. Well, it would help if I had a known working PRI to test and tweak my * box against.. SIGH.. Any help would be greatly appreciated! ___ 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] attension mark spencer
On Tue, 2005-04-12 at 23:16 -0700, amna saleem wrote: hi , I was wondering if i can get some algo or architecture of asterisk...i mean how different channels are working (specially agents,h323)and how call is established... i know i am sounding a bit stupid but i need this ...can you please guide me thanx Yes, it's all in the files ending in .h and .c in /usr/src/asterisk. Now you can do your homework. :) -- Dave Cotton [EMAIL PROTECTED] ___ 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] Codecs and * pass through...
Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are speaking with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ 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] weird call transfer problem
I have the dtmf commented out on sip.conf ;dtmfmode=rfc2833 And the ata have it configured as info The weird thing is tht if I am the one making the call, I CAN do transfers, I just cant make them if I am the one receiving the call. I understand that removing T will forbid the calling user to transfer but as far as I know, I should be able to transfer calls myself... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of El Flynn Sent: Miércoles, 13 de Abril de 2005 12:58 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] weird call transfer problem Anton Krall wrote: Guys. I just had a weird problem. I have my Dial cmd configured with mwtWT as parameters however, a call came in thru a zap channel and I answered on a sip phone. I tried using # as configured on my features.conf file to transfer the call but the transfer prompt never came in, so I asked the person on the zap channel to do the same and voila, he did get the transfer prompt and entered and extension, but what happended is that I was the one that got transfered! Not him! So. Any ideas whats wrong? The sip phone is an ata, a handytone 286 and zaptel cards. Why cant I do the # transfer and they can but Im the one been transfered? The T option allows the *calling* user to transfer the call, which is what happened to you. The t option allows the call recipient to transfer the caller to another extension. So to stop that from happening, remove the T option from the dial command. As to why you yourself can't transfer it might have something to do with the ATA itself, check what dtmfmode is specified in sip.conf. From the sample sip.conf, it says: dtmfmode=info ; either RFC2833 or INFO for the BudgeTone flynn ___ 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 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] Sangoma A101 + Rhino channelbank
Hi, This is exactly what I did - the Sangoma tech support responded fast. They even installed themselves the driver once I gave them access to my machine. I bugged the tech guy (Alex Feldman) for a couple of days, but he acted quite nice trying to solve the problem. It seems it was 2 problems: 1. First it was the t1 cable - both A101 and Rhino channelbank comes with a straight t1 cable, but to connect those two YOU NEED A CROSS CABLE. if you don't have one, you need to manufacture one. Here is the pin-out diagram 1-4 2-5 3-3 4-1 5-2 6-6 7-7 8-8 2. Second, I put only the Sangoma A101 card in the system and loaded the driver. I configured as follows in zaptel.conf span=1,1,0,esf,b8zs fxols=1-24 in zapata.conf signalling=fxo_ls channel = 1-24 NOTE: dont try to configure signalling for each channel because it won't work write first the signalling part (the two lines above) an then you can come with additional things for each channel (like context, callerid etc) I experienced a very strange incompatibility: after making the configuration work, I tried to add a Digium TE100P card in the same computer. Once I add the card in the system, then it is not working anymore. I tried to put the card in each available pci slot, I started only wanrouter and asterisk, without starting the zaptel driver, but still no result. Once I take out the Digium from computer and restart then everything is working. To solve the problem (bcause i need the e1 to connect to telco) I use two machines - one with the Digium and second with Sangoma. I connected the machines with IAX. Felician CHELU, IT Manager Intertel Communications mobile +40 722 552 336 fix +40 21 201 75 29 - Original Message - From: mattf [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 4:34 PM Subject: RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank Keep on bugging the Sangoma guys, I know they are working on several RBS T1 issues right now(They called me Friday to go over a few things) They just need help from users like you and I to find the bugs in their drivers. Have you tried any other signalling types other than LOOP? MATT--- -Original Message- From: Felician CHELU [mailto:[EMAIL PROTECTED] Sent: Monday, April 11, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank Hello, I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with the Rhino fxs chanelbank. Things done: - T1 cross cable = I have carrier, signalling and framnig leds on the channelbank green. - channelbank configuration: t1 - Proto: LOOP Frame: esf Clock: slave Coding: b8zs channels(analog) : Function:A-fxsMode:loop - zaptel.conf span=2,1,0,esf,b8zs fxols=32-55 (i have a span 1 with a digium e1) - zapata.conf signalling=fxo_ls - wanpipe1.conf [devices] wanpipe1 = WAN_AFT, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 2 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= MASTER ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO LBO = 0DB INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 2 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO I already called Sangoma and Rhino support, but after hours of long distance call conversation the problem is still not solved. Finnaly, a guy from Rhino told me that their asterisk expert (which was not avaliable) knows about this problem and that it is that the sangoma driver is not communicating with asterisk. The wanrouter starts ok, after ztcfg I see the channels configured. The problem: i don't have dialtone on phones. Question: When i enter zttoll, if i go to the sangoma span and I make loop then it freezes. Is it normal? If someone has experienced this combination and made it work please give me a sign. Thank you. PS: Felician ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Codecs and * pass through...
Etienne, howzit I am not 100% sure about this, but Net2phone do not always use standard SIP as the protocol. They have their own proprietry protocol as well, so perhaps your phone is trying to talk on the proprietry protocol. For G723.1 passthrough, you just allow it, and it should work fine, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote: Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are speaking with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ 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 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] Sipura SPA-841 and Asterisk 1.0.7 with chan_misdn
HI Everyone, I have run into a rather unusual Problem.. My Config as follows System 2.6.9 Kernel mISDN 0.0.3.RC6 AVM Fritz! X 3 chan_misdn-0.1.0 Asterisk CVS Stable. Handsets: Micronet SP5100 Micronet SP5001 ATA Sipura-841 (Latest FIrmware) When I Make Calls from the SPA to PSTN(or the reverse), at first calls go through clear. After the Second or third Call, we wind up with 4-7ms jitter. If I transfer the call to the Micronet(which doesn't seem to experience ANY difficulties), call is cleartransfer back to the SPAjitter again the Jitter is only heard on the SPA end...the PSTN end of the call is fine Calls from sip to sip present no issues, as with calls to IAX2 trunks. Has anyone else run into this difficulty, or at least point me in a direction to try and fault find this Much Thanx Dave ___ 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] Codecs and * pass through...
Clive, cool - winter is getting quite near ova here... Well, how would I find out what is happening - I mean how do I know what * is connecting with to net2phone. "...They have their own proprietry protocol..." I thought it was because of the G723.1 codec and passthrough - but the I must take the voice prompts way. :-) (Didn't thought that it'll cause a problem - just the warnings and notices but continue still...) Thank you for that tip. "...For G723.1 passthrough, you just allow it..." --- So that is in "sip.conf" [general] disallow=all; allow=G723; allow=ulaw; allow=alaw; allow=gsm; (some text later) [net2phone] (some text) canreinvite=yes; (some text) --- Sources for net2phone: http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Net2phone http://www.voip-info.org/tiki-index.php?page=Net2phone PS - I do get a frame error about expecting 4 getting 256 when * is trying to initiate to call through to net2phone device MAX IP-10 through the net2phone network - could be that protocall you were talking about or have I completely missed the plot? Kind Regards Etienne [EMAIL PROTECTED] wrote: Etienne, howzit I am not 100% sure about this, but Net2phone do not always use standard SIP as the protocol. They have their own proprietry protocol as well, so perhaps your phone is trying to talk on the proprietry protocol. For G723.1 passthrough, you just allow it, and it should work fine, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote: Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are "speaking" with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ 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 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 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] How to get list of codecs
Will try, thanks :) Pavel Siderov - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 6:45 PM Subject: Re: [Asterisk-Users] How to get list of codecs mmm i think Agi by itself does not provide a way to do so. And the codecs are negotiated depending upon the codec that both call sides support. So, i belive that the only way is making your own implementation of AGI in res_agi.c :) Hopefully someone will come up with a better idea :-) best regards On Apr 12, 2005 1:34 PM, Pavel Siderov - Hostmates [EMAIL PROTECTED] wrote: Hi Guys, Is it possible to get the UAC supported codec list when making a call. I want to assign to variable1 and variable2 the first 2 supported codecs using AGI script e.g. $variable1=g723 $variable2=g729 Somebody can help me ? Any help is appreciated. Thanks, Pavel Siderov ___ 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 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 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] Re: Problem reading digits from OH323 caller
In article [EMAIL PROTECTED], Joe S [EMAIL PROTECTED] wrote: I am setup SJPhone and called the voicemail, but Asterisk cannot collect the mailbox number and password. Tried it also with Netmeeting with no luck. Does anyone knows something about this? Try experimenting with the inBandDTMF and userInputMode settings in the oh323.conf file. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] PortaSIP/PortaBilling incompatibility (provider: sipcall.ch)
On Mon, Apr 11, 2005 at 08:18:43PM +0200, gramels wrote: If Useragent field in this config corresponds to User-Agent field in Asterisk's SIP messages and you may change it to something that doesn't contain a word Asterisk - please try to do so; in such case PortaSIP will not apply remote IP auth. I might have a similar problem with SER (sipphone.com) and my Asterisk. However the mentionned work around doesn't work. Funnily the register works with the same password. What happens: Apr 13 09:34:45 NOTICE[2495]: chan_sip.c:6831 handle_response: Failed to authenticate on INVITE to '17476691152 sip:[EMAIL PROTECTED];tag=as41277c10' log: Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: portasipfriendly Date: Wed, 13 Apr 2005 07:34:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 343 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84ac3c477a344ab166ec9 Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1706 req_src_ip=80.83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e.1876 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: portasipfriendly Content-Length: 0 Reliably Transmitting: Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: portasipfriendly Authorization: Digest username=17476691152, realm=sipphone.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=425cc84ac3c477a344ab166ec9 7f6efdadcc6bb3, response=1bdef16f3de89e9194116a2a0135a495, opaque= Date: Wed, 13 Apr 2005 07:34:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 343 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. bd57 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84bb62fad894c1533475c 08dead3e27baa5 Content-Length: 0 Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1707 req_src_ip=80.83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= sip:[EMAIL PROTECTED] via_cnt==1 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. bd57 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: portasipfriendly Content-Length: 0 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK02c48610 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: portasipfriendly Authorization: Digest username=17476691152, realm=sipphone.com, algorithm=MD 5, uri=sip:[EMAIL PROTECTED], nonce=425cc84bb62fad894c1533475c 08dead3e27baa5, response=219b7fec33546a32a830edfba25fa601, opaque= Date: Wed, 13 Apr 2005 07:34:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Sip read: SIP/2.0 401 Unauthorized To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84bb62fad894c1533475c Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1702 req_src_ip=80. 83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= Transmitting: To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. 6caf Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: portasipfriendly Content-Length: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Cisco 7960s and skinny
On Tue, Apr 12, 2005 at 04:38:10PM -0500, Andy Hamilton arranged a set of bits into the following: Simon: I have had Skinny going on a 7960 (which I then reimaged to SIP). I currently run a 7910 on Skinny (using chan_sccp) and use the aforementioned 7960 simultaneously. Since you mentioned that you will have 50 phones, I assume you are using them in a business setting. I would *highly* recommend using SIP, as I have found that the skinny driver is not as reliable as it could be (not criticizing Jan or Julien at all, here). Even if you were, my own view is that chan_sccp is probably not the thing to run on a client's PBX (not sure how good chan_skinny is, didn't work the first time I tried which is why I do chan_sccp). My own personal one, yes, a business where I worked full time and had safe_asterisk or similar working, perhaps, anywhere else no. My biggest task is getting in some of the big bugfixes and bad behavior fixes that have been major issues. In testing at the moment is a fix to allow speeddials to work at any time (meaning you could in theory create a speeddial that auto-navigated a remote IVR), instead of crashing if the handset was up. My next task is to get subscribe/notify working (if anyone has looked at this code could they drop me a few pointers), which should be pretty easy. Another thing which I might do is implement a live/hot keypad so any keypress triggers a call, some people seem to like this, but I personally can't stand it. (In any case it should be a 5 line patch if enabled all the time closer to 50 lines when you have a per-device config option. Also I've finally updated the web site to clean it up and hopefully add some more info. Reimaging the 50 of them should only take a while (depending on what version of CCM they have at the moment). I reimaged 12 phones once for a business and it took less than 30 minutes after I got it going (toying with the phones to get them to take the image, exactly how the config files were to be set up, etc...). I imagine you could easily get the whole thing done in less than a day (reimaging and config files), then figure out your dialplan. Then there is the whole issue of writing the config files...but you'd have to do those with Skinny, anyhow. I think with SIP you'll have much better reliability. -Andy FWD: 428725 On Apr 12, 2005 12:48 PM, Morris, Simon [EMAIL PROTECTED] wrote: Hello, Does anyone else have * running with Cisco 7960 phones and skinny? All the advise I am reading so far is telling me to load the SIP image on the phone but I'd like to know what I'm going to lose by persisting with skinny (Not reimaging 50 phones is one benefit amongst others of skinny) Thanks for any comparisons you can provide Rgds ~sm ___ 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 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 pgpIcQ4U3tfYJ.pgp Description: PGP signature ___ 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] iaxcomm
Hi! I was using iaxcomm but due to some reason am not able to transfer calls to some other extensionwhat maybe the problem do i have to make some changes to my extensions.conf??or iax.conf to be able to transfer calls Thanks Amna ___ 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] Turtle Firewall - Sip user
Hi, I have a Turtle firewall separating public and private address.I need a sip user "SJPhone" on a private address to connect to a public Asterisk server.Im a bit confused about what to solution to follow from the wiki's, NAT tunnel etc. If anyone can give me aadvise. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Fritz and TDM400
Is it possible to have on the same machine an ISDN Fritz Card and a TDM400 with two FXO ports? If so, is there any place I can find instructions to configure it? ___ 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] ISDN Fritz and TDM400
Is it possible to have on the same machine an ISDN Fritz Card and a TDM400 with two FXO ports? If so, is there any place I can find instructions to configure it? There should be nothiong special in using two cards. Just insert both cards into different slots an configure each card according to the instructions. In the dialplan you have to specify which card to use to dialout and there to forward incoming calls. Elmar ___ 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] ISDN Fritz and TDM400
Thanks for your reply, my doubt rest on the fact that there are two ways of configuring it: One using the Bristuff from Junghanns and the other using CAPI. Is there any major difference/advantages to one or the other? p.s. I cant find instructions on how to configure bristuff besides what comes with the package. ___ 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] codec quality
Hi there I am using Meetme and have been testing with 2 different codecs GSM and g.711 these seem to be the only 2 free codecs which are supported by my soft phones (built using the RTC Client API). All users will be using this same softphone when communicating. The quality of g.711 (ulaw) I have found to be good, but it uses too much bandwidth. Although it sends less data, the quality of GSM is not great it is quite fuzzy and not pleasant. Is there any way to improve the quality of this codec? Or perhaps it is just an inferior codec to others which transmit at 13kbps or less (such as g.729 or ilbc)? Skype uses ilbc and the quality seems really good. Lastly, what is the overhead that is added onto the audio packets? For instance, GSM (13 kbps) sends at about 40 kbps and g.711 (64 kbps) sends at about 80 kbps? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New PRI install with new te110p
On April 13, 2005 02:40 am, Me wrote: == Primary D-Channel on span 1 down Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No D-channels available! Using Primary on channel anyway 24! The telco hasn't turned up your D channel yet. If I change signalling to pri_net the errors go away, either way I can receive calls into Asterisk. When you're pri_net you are creating the D channel, but if you're connected to a telco there is no way you'd receive calls in this configuration. How should the signalling be set, to cpe or net? You're the CPE. Any idea what's causing this error? I am not entirely sure my PRI is 100% up even, * seems to be talking to it because when I pull the cable it starts giving me alerts and such, the alerts go away when I plug the cable back in. The T1 is likely up, which is what makes the LED on the back go green. When you pull the cable, the T1 is down and Asterisk tells you this. PRI is signaling on top of the T1. You can have the T1 up and have no D channel. Wait for your telco to tell you the PRI is provisioned and up (they usually work with you on the phone while they provision it, because there are a few test calls made and so on). Now if you are able to receive calls into asterisk in this state... then colour me confused. -A. ___ 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] invalid extension (need help)
On April 13, 2005 12:35 am, amna saleem wrote: I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup You've already answered the call; no need to answer again, although it won't hurt. Make sure that these lines are either in the same context that your call is executing within, or that it is included in that context. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 12, 2005 11:36 pm, Kevin P. Fleming wrote: Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per Yes, but then what are you doing with it? You're shuttling the new data to/from a network card in a lot of cases. Combined with other traffic over the PCI bus for normal system operation I could see you coming close to the limitations of regular ole PCI. second of traffic. People looking a DS3 cards are also likely to deploy them in servers with multiple independent PCI buses, which would then allow for even more bandwidth. The mind boggles at the possibilities! True enough, but you still need to marshall the data going between PCI busses and to system memory. Certainly not impossible problems to overcome but they do add to the fun of getting a low latency VOIP system together. -A. ___ 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] Version 0.80 of IPS released
I removed the old version, deleted the install directory, and installed a new version, then changed the config so it connects to a different pbx. I am still seeing all the old extensions, nothing from the new pbx, even though ipswitchboard is connecting to the new pbx. Where are the extensions being cached and how do I do a completely new install? Can you add the ability to switch pbx's? New PBX: == Manager 'anguilla' logged off from 206.48.59.5 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'anguilla' logged on from 206.48.59.5 Chris Mason www.anguillaguide.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] Re: Cisco 7960s and skinny
My biggest task is getting in some of the big bugfixes and bad behavior fixes that have been major issues. In testing at the moment is a fix to Yes, I'm using * in a business environment with cisco 7960 and 7905 phones. Sip is the more stable solution. well no busy status line 'cause the cisco sip firmware does not support it. I was testing your chan_sccp. It's under development and I got some crash or phones issue, but I think sccp could be the best flexible system for a PBX. In my spare time I'm working on your chan_sccp code to understand how to get customized and localized (I'm in Italy) softkeys. ___ 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 / Quintum CRSP codec problems
Hi Guys, I have following scenario which causes an issue related to codecs (please look below)[asterisk] - Quintum CRSP* / Quintum CMS - PSTN * Quintum Call Relay SP (CRSP - http://www.quintum.com/main/servproducts.html?id=15), Quintum CMS - H323 basedgatewayWhen a call is being placed using a SIP client,UAC (sip client)sends a list of supported codecsto theasterisk and combinedlist isbuild as supposed, e.g. Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) My Asterisk has following codec settings applied: disallow=allallow=g723allow=g729allow=alawallow=ulaw Asteriskforwarding calls with prefix 00 to Quintum SIP-PSTN (Quintum CRSP) gatewaywhich has enabled g723,g729, ulaw and alaw codecs - the sameas ones on the asterisk side. Quintum CRSP sends back only the first codec out of the supported codecs list on my side (asterisk) and within the combined listthere isonly the first codec on asterisk side. Please find below log out of described above behaviour: Sip read:SIP/2.0 200 OKCSeq: 102 INVITECall-ID: [EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Content-Type: application/sdpFrom: "pavel"sip:[EMAIL PROTECTED]:5060;tag=as74cec9dbTo: sip:[EMAIL PROTECTED];tag=3ef4af85-1112bVia: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK5568aa34Content-Length: 168User-Agent: Quintum/1.0.0 v=0o=Quintum 33034 31527 IN IP4 1.2.3.4s=VoipCallc=IN IP4 1.2.3.4t=0 0m=audio 10858 RTP/AVP 4c=IN IP4 1.2.3.4a=rtpmap:4 g723/8000/1 10 headers, 8 linesFound RTP audio format 4Peer audio RTP is at port 1.2.3.4 :10858Found description format g723Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) where us is asterisk, peer is the Quintum CRSP This way if the SIP client supports only G729 the call fails since the combined list of codecs would be again g723 as long as the g729 is not the first in the list of asterisk's codecs. Is there any way I can forcably determine the codecs reported for the peer out of the asterisk's codec capabilities list? Any idea how I could force the asterisk to change the order of supported codecs according to the SIP client first codec in the list or how I could force the Quintum tosend back the full list of supported codecs ? Thanks and regards, Pavel ___ 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] Compile/modprobe issue
Thank you very much. That has done it :-) Jeffrey C. Ollie wrote: On Tue, 2005-04-12 at 20:29 -0700, Steven P. Donegan wrote: I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux (2.6.10 kernel patched as suggested). I get compile warnings and modprobe failure on zaptel stuff: zaptel: Unknown symbol crc_ccitt_table I'm assuming that something needs to be in the kernel space that isn't - any pointers to resolving this would be appreciated. You need to have: CONFIG_CRC_CCITT=m set in your kernel config. ___ 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 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 7940G SIP Conversion
Hi, I have three Cisco 7940G phones that I'm trying to convert to SIP Image P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows XP platform. I have configured my DHCP server to hand out the correct TFTP address as the phone confirms it knows where to find a TFTP server. In the Cisco TFTP status window, I'm receiving the following message continuously: Wed Apr 13 08:21:44 2005: Sending 'OS79XX.TXT' file to 192.168.1.30 in binary mode# I would expect it to attempt to load the image file next that is listed in the OS79XX.txt file (P003-07-4-00, but it continues to load JUST the OS79XX.TCT file continuously. Any ideas? Michael J. West [EMAIL PROTECTED] WESTMark Consulting, Inc. 34 Wasilla Drive Worcester, MA 01604-2411 ___ 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] Multiple TDM400x Cards on the same box
Hi All, Has anyone installed multiple TDM cards on the same box? I'm trying to run such a configuration With [EMAIL PROTECTED], and it fails for some reason. Any pointers ? Nir S ___ 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] Question about Routing Order in .conf files
The question is in the
logical route that asterisk takes when reading and executing the scripts.
Please see the (?) questions beside the lines.
The goal is not to comment
the lines exten = snip in the [ext-local] everytime that I make
a change using the AMP GUI. Also it would be nice to be able to give priority
to the *_custom.conf if possible.
Thank you in return.
Extensions_additional.conf
[aa_1]
include = aa_1-custom
exten =
1,1,Goto(ext-local,7726258,1) ;
exten =
2,1,Goto(ext-local,7726259,1) ; this take the call to the
[ext-local]
exten =
3,1,Goto(ext-local,7726257,1) ;
exten =
fax,1,Goto(ext-fax,in_fax,1) ;
[ext-local]
include =
ext-local-custom ; ? should this not be held in priority first over any of the
contents in [ext-local] ?
exten =
7726257,1,Macro(exten-vm,[EMAIL PROTECTED],7726257)
exten =
7726258,1,Macro(exten-vm,[EMAIL PROTECTED],7726258)
? I am able to
monitor the call if I comment the line out ? as it then seems to go
to the [ext-local-custom] located in the extentions_custom.conf ?
;exten =
7726259,1,Macro(exten-vm,[EMAIL PROTECTED],7726259)
;exten =
7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
;exten =
7726259,2,SetVar(CALLTIME=${DATETIME})
;exten =
7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
;exten =
7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
;exten =
7726259,5,DIAL(SIP/7726259,15,t) ;exten =
7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
exten =
9873022,1,Macro(exten-vm,[EMAIL PROTECTED],9873022)
extentions_custom.conf
[ext-local-custom]
;test to see if this stays
exten =
7726259,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten =
7726259,2,SetVar(CALLTIME=${DATETIME})
exten =
7726259,3,SetVar(CALLPATH=/var/www/html/monitor/7726259)
exten =
7726259,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
exten =
7726259,5,DIAL(SIP/7726259,15,t) exten =
7726259,6,Macro(exten-vm,[EMAIL PROTECTED],7726259)
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Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards
The only time PLC makes sense is thwn you are converting FROM VoIP to something else. So PLC would be done on chan_sip or chan_IAX, or chan_h323 on the receiving end. This is for 1.0.x. For CVS-HEAD you would want to do this on the receiving side in the PLC stuff. parijat wrote: Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Busy line status and chan_capi?
What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Kib ___ 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] x-ten lite error
Hallo, I have just set up [EMAIL PROTECTED] and configured two extensions, 200 and 201 with x-ten lite. Ext 201 seems to be ok (i tried to dial 200 and the system answered that the extension was not available atg the moment and let me leave a voicemail message), Ext 200 seems to be blocked: for any number that i try to dial I always receive the busy tone and the message Call failed: 403 forbidden appear on the softphone screen. Any suggestion? tia brgs Francesco ___ 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] VAD/DTX implementation through zaptel cards
Pls could u be more elaborate as I am new to asterisk.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, April 13, 2005 7:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards The only time PLC makes sense is thwn you are converting FROM VoIP to something else. So PLC would be done on chan_sip or chan_IAX, or chan_h323 on the receiving end. This is for 1.0.x. For CVS-HEAD you would want to do this on the receiving side in the PLC stuff. parijat wrote: Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 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] 3-Way Calling in Asterisk
You just described a conference call which is supported by most phones. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of aram Sent: Tuesday, April 12, 2005 6:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] 3-Way Calling in Asterisk Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 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] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Cisco 7940G SIP Conversion
I made the same mistake with my 7960 The content of 'OS79XX.TXT' should be P0S3-07-4-00 and not P003-07-4-00 Same goes for SIPdefault.cnf. After the change everything worked like magic -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael West Sent: Wednesday, 13 April 2005 22:27 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940G SIP Conversion Hi, I have three Cisco 7940G phones that I'm trying to convert to SIP Image P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows XP platform. I have configured my DHCP server to hand out the correct TFTP address as the phone confirms it knows where to find a TFTP server. In the Cisco TFTP status window, I'm receiving the following message continuously: Wed Apr 13 08:21:44 2005: Sending 'OS79XX.TXT' file to 192.168.1.30 in binary mode# I would expect it to attempt to load the image file next that is listed in the OS79XX.txt file (P003-07-4-00, but it continues to load JUST the OS79XX.TCT file continuously. Any ideas? Michael J. West [EMAIL PROTECTED] WESTMark Consulting, Inc. 34 Wasilla Drive Worcester, MA 01604-2411 ___ 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 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ 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] Re: Cisco 7960s and skinny
On Wed, Apr 13, 2005 at 01:47:22PM +0200, Sergio arranged a set of bits into the following: My biggest task is getting in some of the big bugfixes and bad behavior fixes that have been major issues. In testing at the moment is a fix to Yes, I'm using * in a business environment with cisco 7960 and 7905 phones. Sip is the more stable solution. well no busy status line 'cause the cisco sip firmware does not support it. I was testing your chan_sccp. It's under development and I got some crash or phones issue, but I think sccp could be the best flexible Please, and this goes for all chan_sccp users run asterisk with the -g option to get coredumps if it crashes, and send me the backtrace (NOT the coredump). system for a PBX. In my spare time I'm working on your chan_sccp code to I agree, sccp or a similar protocol is great as it allows the PBX to contain most of the features that usually go to the phones, allowing an amazing flexibility. understand how to get customized and localized (I'm in Italy) softkeys. I'm not sure what if anything there is to localize, IIRC chan_sccp transmits no text to the user except for softkey names, and their you might be out of luck. Hope to see some Aussie Asterisk users at LCA! Julien PS: I'm now starting to look at writing a basic implmentation of CDP for setting vlan's on cisco phones, expressions of intrest wanted! pgpzUn453vN8v.pgp Description: PGP signature ___ 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] Who is willing to help an Asterisk newby?
As of last night, I have a working Asterisk system, courtesty of [EMAIL PROTECTED]. Now comes the need to iron out the wrinkles and fine-tune my setup. Who would be willing for me to shoot questions at him/her which would just annoy the list if I brought them here? Here is my setup: P3/450Mhz, 256MB RAM, 80GB Disk (Compaq Deskpro EN), 1 X100P OEM FXO connected to PSTN (Telekom Austria) 1 Sipura 2000 connected to a Siemens Gigaset DECT/GAP Cordless unit Multiple X-lite Softphones Broadvoice configured as a trunk FWD configured as a trunk Here's one of my first questions: The system seems to be re-registering with Broadvoice every 20 seconds or so. Things work, but this seems to be an awful lot of unnecessary activity both on the network, and in the logfile. Here is how it manifests in the logfile: Apr 13 15:18:44 DEBUG[28010]: Registration successful Apr 13 15:18:44 DEBUG[28010]: Cancelling timeout 13449 Apr 13 15:19:00 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:00 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:00 DEBUG[28010]: Scheduled a registration timeout # 13452 Apr 13 15:19:00 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 144: Found Apr 13 15:19:00 DEBUG[28010]: Registration successful Apr 13 15:19:00 DEBUG[28010]: Cancelling timeout 13452 Apr 13 15:19:16 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:16 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:16 DEBUG[28010]: Scheduled a registration timeout # 13455 Apr 13 15:19:16 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 145: Found Apr 13 15:19:16 DEBUG[28010]: Registration successful Apr 13 15:19:16 DEBUG[28010]: Cancelling timeout 13455 Apr 13 15:19:32 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:32 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:32 DEBUG[28010]: Scheduled a registration timeout # 13458 Apr 13 15:19:32 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 146: Found Apr 13 15:19:32 DEBUG[28010]: Registration successful Apr 13 15:19:32 DEBUG[28010]: Cancelling timeout 13458 Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:19:49 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:49 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:49 DEBUG[28010]: Scheduled a registration timeout # 13461 Apr 13 15:19:49 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 147: Found Apr 13 15:19:49 DEBUG[28010]: Registration successful Apr 13 15:19:49 DEBUG[28010]: Cancelling timeout 13461 Apr 13 15:20:05 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:20:05 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:20:05 DEBUG[28010]: Scheduled a registration timeout # 13464 Apr 13 15:20:05 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 148: Found Apr 13 15:20:05 DEBUG[28010]: Registration successful ___ 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] Who is willing to help an Asterisk newby?
On Wed, 2005-04-13 at 14:21 +0100, Wolf N. Paul wrote: As of last night, I have a working Asterisk system, courtesty of [EMAIL PROTECTED]. Now comes the need to iron out the wrinkles and fine-tune my setup. Who would be willing for me to shoot questions at him/her which would just annoy the list if I brought them here? Personally I think it is much more beneficial for you to ask the list questions - rather than disappear into a one-on-one session with someone. Not only for yourself who would get a wider range of advice and opinions from various people on-list, but for me :) who doesn't know all that much about * And of course for anyone reading the archives in time to come... ~sm ___ 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] 3-Way Calling in Asterisk
i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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] Who is willing to help an Asterisk newby?
You can turn off the amount of logging in the log.conf setting. As far as the registration goes .. that would be under your Sipura Settings. You may only want to reduce this to 60 sec registration .. I find that any longer sometime effects longevity of server to find you in the route. Only my 2cents ... and I am no expert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolf N. Paul Sent: Wednesday, April 13, 2005 8:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Who is willing to help an Asterisk newby? As of last night, I have a working Asterisk system, courtesty of [EMAIL PROTECTED]. Now comes the need to iron out the wrinkles and fine-tune my setup. Who would be willing for me to shoot questions at him/her which would just annoy the list if I brought them here? Here is my setup: P3/450Mhz, 256MB RAM, 80GB Disk (Compaq Deskpro EN), 1 X100P OEM FXO connected to PSTN (Telekom Austria) 1 Sipura 2000 connected to a Siemens Gigaset DECT/GAP Cordless unit Multiple X-lite Softphones Broadvoice configured as a trunk FWD configured as a trunk Here's one of my first questions: The system seems to be re-registering with Broadvoice every 20 seconds or so. Things work, but this seems to be an awful lot of unnecessary activity both on the network, and in the logfile. Here is how it manifests in the logfile: Apr 13 15:18:44 DEBUG[28010]: Registration successful Apr 13 15:18:44 DEBUG[28010]: Cancelling timeout 13449 Apr 13 15:19:00 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:00 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:00 DEBUG[28010]: Scheduled a registration timeout # 13452 Apr 13 15:19:00 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 144: Found Apr 13 15:19:00 DEBUG[28010]: Registration successful Apr 13 15:19:00 DEBUG[28010]: Cancelling timeout 13452 Apr 13 15:19:16 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:16 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:16 DEBUG[28010]: Scheduled a registration timeout # 13455 Apr 13 15:19:16 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 145: Found Apr 13 15:19:16 DEBUG[28010]: Registration successful Apr 13 15:19:16 DEBUG[28010]: Cancelling timeout 13455 Apr 13 15:19:32 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:32 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:32 DEBUG[28010]: Scheduled a registration timeout # 13458 Apr 13 15:19:32 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 146: Found Apr 13 15:19:32 DEBUG[28010]: Registration successful Apr 13 15:19:32 DEBUG[28010]: Cancelling timeout 13458 Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:19:42 DEBUG[28010]: Manager received command 'Command' Apr 13 15:19:49 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:19:49 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:19:49 DEBUG[28010]: Scheduled a registration timeout # 13461 Apr 13 15:19:49 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 147: Found Apr 13 15:19:49 DEBUG[28010]: Registration successful Apr 13 15:19:49 DEBUG[28010]: Cancelling timeout 13461 Apr 13 15:20:05 DEBUG[28010]: # Testing 147.135.4.128 with 192.168.1.0 Apr 13 15:20:05 DEBUG[28010]: Target address 147.135.4.128 is not local, substituting externip Apr 13 15:20:05 DEBUG[28010]: Scheduled a registration timeout # 13464 Apr 13 15:20:05 DEBUG[28010]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 148: Found Apr 13 15:20:05 DEBUG[28010]: Registration successful ___ 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 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 on debian sarge doesn't start with CAPI module errors
Hello, Fresh install of Debian Sarge and asterisk from the debian archives. Asterisk doesn't start and dies with the following message. [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Apr 13 15:38:44 NOTICE[1580]: chan_capi.c:2635 load_module: CAPI not installed! Apr 13 15:38:44 WARNING[1580]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Apr 13 15:38:44 WARNING[1580]: chan_capi.c:2811 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Apr 13 15:38:44 WARNING[1580]: loader.c:440 load_modules: Loading module chan_capi.so failed! I have an ISDN card I'm going to install later, but I want to get Asterisk up and running with SIP first. lon0asterisk01:~# dpkg -l | grep asterisk ii asterisk 1.0.5-2open source Private Branch Exchange (PBX) ii asterisk-app-d 0.0.20050203-2 Text entry application for Asterisk ii asterisk-app-f 0.0.20050203-2 Softfax application for Asterisk ii asterisk-chan- 0.3.5-11 Common ISDN API 2.0 implementation for Aster ii asterisk-confi 1.0.5-2config files for asterisk ii asterisk-dev 1.0.5-2development files for asterisk ii asterisk-doc 1.0.5-2documentation for asterisk ii asterisk-gtk-c 1.0.5-2gtk based console for asterisk ii asterisk-h323 1.0.5-2asterisk H.323 VoIP channel ii asterisk-promp 1.0-1 German prompts for the Asterisk PBX ii asterisk-promp 0.0.20040928-1 French voice prompts for Asterisk ii asterisk-promp 0.8-2 Swedish voice prompts for Asterisk ii asterisk-sound 1.0.5-2sound files for asterisk ii asterisk-web-v 1.0.5-2web based (GCI) voice mail interface for ast lon0asterisk01:~# dpkg -l | grep capi ii capisuite 0.4.5-2easy fax and voice box solution for ISDN/CAP ii libcapi20-23.6.2005-01-03 libraries for CAPI support Any ideas? Thanks ~sm ___ 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] ni1 (ppp) and national isdn on te110p
I was wondering how to divide channels into data and voice in zapata.conf and zaptel.conf. I have a PRI line. On half the channels I would like to set up a direct dial to one of our clients ISDN modems (done via ni1) so that I can provide internet access. I would like the other half to handle the DID numbers so that Asterisk can make phone calls. How do I go about doing this? Thank you, Jason McAffee The Technology Group ___ 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] Re: Running asterisk without special hardware
Damian Funnell [EMAIL PROTECTED] writes: Hi Manish, Sure can, although you will need a timing source. Not necessarily. In a pure VoIP environment, I don't know of any asterisk application which needs timing other than meetme. I.e. if you need conferencing, you'll need ztdummy as a timing source. If not, you can just download * 'as is', compile and install it into some place, and finally set up your dial plan. That's it. Please read the Wiki for details on * setup and ztdummy/timing as well. All this info is readily available there, and in detail, too. Regards, Bruno. ___ 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] Acceptable voice time delay
chawki, If I may answer this; If you have 600ms round trip to voipjet, I would guess there are further problems with your line than simple latency. That additional 2.5 seconds of delay may be any combination of things, but I would look first to your ISP and their backbone. I have tried voipjet, and while I wasn't enamored with it, I did not find any latency issues that you speak of. Good luck! Sean chawki hammoud wrote: thank you Rob: the problem is that I am experiencing about 3 secs latency although the ping is 600ms which is a round trip packet travel time. so i should experience about half a sec latency including the voipjet server response and the latency to the pstn. that is annoying, but nothing compared to about 3 sec. do you think the rest of the delay is due to voipjet slow response to the pstn network or some other issues would you be bale to clculate where the 3 sec is comming from thanks. Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. ___ 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] Question about Macros
Can you post here what is working, for the benefit of everyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe S
Sent: Wednesday, April 13, 2005 12:38 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Question about Macros
Hi all,
Never mind, I solved the problem with Read...
Joe
From: Joe S [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Question about Macros
Date: Tue, 12 Apr 2005 21:39:31 -0500
Hello,
I am trying to build a macro + menu together. Here is the idea:
Dial 1000 to get to conference, menu comes, press 1 to login as admin
or press 2 to login as user.
Here is my code in extensions.conf:
exten = 1000,1,Macro(conf2me,1000)
[macro-conf2me]
exten=s,1,Wait,1
exten=s,2,Answer
exten=s,3,DigitTimeout,5
exten=s,4,ResponseTimeout,10
exten=s,5,Background,welcome-instructions
exten=i,1,Playback,pbx-invalid
exten=i,2,Goto,s|5
exten=t,1,Hangup
exten=1,1,MeetMe(${ARG1}|ipda)
exten=2,1,MeetMe(${ARG1}|ipd)
It doesn't work, it is giving the error that '1' is not defined in the
default context, probably it exited the menu (macro).
Thanks,
Joe
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[Asterisk-Users] PCI 1xE1, 2xE1 cards from Russia for MFC/R2 signaling for Asterisk IP-PBX
Hello All. If you are intrested in subj drop me e-mail for additional info. Info in english will be available soon. ICQ: 172468035 MSN: litnimax(at)hotmail.com (do not send mail here!) e-mai: litnimax(at)asterisk-support.ru - - Maxim Litntisky Head of Telecom Department Key Solutions Russia Moscow http://www.ksolutions.ru http://www.asterisksupport.ru ___ 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] DS3000P - 20 E1 capacity on single card
Matt Klein wrote: Kevin, Mmm. Yep. -m On Tue, 12 Apr 2005, Kevin P. Fleming wrote: Matthew Boehm wrote: So, no hardware encoding on this beast? The announcement on the website makes no mention of transcoding, echo cancellation or toast-and-jam making, so at this time, no, there is no hardware transcoding apparently included. (Besides, would you really want a board that could only ENcode? G) Since encoding typically requires 5 times as much compute as decoding, for CELP based codecs, an encode onyl board would not be as dumb as it seems at first sight :-) Regards, Steve ___ 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] SNOM 220 with 7 lines
I have a SNOM 220 with a 20-button sidecar. The configuration for the five lines (buttons) on the main phone is straigth forward: display name, account, password, registrar. I would like to get each of the sidecar buttons to register with Asterisk in Line mode so that I can have incoming calls from each DID number appear on a separate button. For the sidecar buttons, however, all I have is a URL. I've tried sip:[EMAIL PROTECTED] which won't work because there's no password. I've tried sip:421:[EMAIL PROTECTED], but the : is converted into a %3a and it doesn't register. \: doesn't work either. Does anyone have any insight? Thanks, ___ 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] Zaptel and Fritz Card
Does anyone has instructions on how to install the Fritz PCI Card with Zaptel? There is no clear instructions in Junghanns.net nor on the Fritz Card ___ 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] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
Try the 0.7.2-pre1 version of asterisk-oh323. It can be found at the Download section on the home page of asterisk-oh323. Michael. Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO ___ 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] Transferring a call
Title: Transferring a call Hello, I have successfully connected an Asterisk PBX to an old Panasonic Phone System using an AVM Fritz PCI card. But when I make a call through the Asterisk PBX to the old phone system, and the receiver wants to transfer the call to another internal number, I get a busy tone. Does anyone have any suggestions to overcome this problem? Any help would be great. Thanks, Dennie __ This mail has been scanned for viruses by an AXS Web Firewall, powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.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
Re: [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'
NVC List Manager wrote: On Friday 08 April 2005 11:57, Ronald Wiplinger wrote: What does it mean, and how can I fix it? Use a browser and turn off the Publish request on the Advanced page. (Obviously you turn the browser to the IP of the phone. See Snom manual for more help.) I looked up the word publish in the manual, but it does not give me a clue: Publish Presence Control the presence status information through this setting. *CLI shows: Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' bye Ronald ___ 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] Zaptel and Fritz Card
Robson Ribeiro wrote: Does anyone has instructions on how to install the Fritz PCI Card with Zaptel? There is no clear instructions in Junghanns.net nor on the Fritz Card Do you want to install the Fritz! Card only, or in conjunktion with a Zaptel card? If you only want the Fritz! card, only chan_capi is need from junghanns.net If you want to install a zaptel card as well you will need an additional driver for the zaptel card you have. If it is a HFC ISDN card, you must download the bristuff of junghanns.net in addition to the chan_capi. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ 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] Local Echo
Hi, It's not probable that the delay is just in the sidetone. It is more probable that the echo is caused by reflected energy somewhere very far downline, perhaps even at the far terminating end of the call. Yes, I know that the person at the far end of the call does not hear an echo, but that doesn't mean that the far end is not the cause of the problem. Think about this: If I scream and the energy is reflected by the wall of a building, I will hear an echo. If you stand at that wall and listen, you will only hear me scream but not the echo. You will never hear an echo at the point where the reflection occurs because there will be no delay at that point. You will only hear an echo somewhere away from the reflection point so that the signal will have some time delay caused by the travel path. If you hear an echo, the reflection is somewhere away from you so that there can be a delay caused by the signal going somewhere and then coming back. Try this experiment: call a number that is answered directly by the asterisk box and see if you get the echo. If you do, it is definitely a local problem. Regards, Neal -Original Message- From: Adam Goryachev [SMTP:[EMAIL PROTECTED] Sent: Tuesday, April 12, 2005 9:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [Asterisk-Users] Local Echo On Tue, 2005-04-12 at 17:13 -0700, Noah Silverman wrote: Thanks Jeff, Your explanation helps. You are correct. There is delay in the sidetone. It annoys me, but the other party doesn't her it. (You're right that the other party is on a POTS line.) I assume that the echo must be between the SIP phone and Asterisk. Since the actuall call sounds fine to both me and the other party, then the Zapata stuff must be working fine. Right?? No, thats what everyone keeps telling you. Everything is working fine on both ZAP and SIP sides, just that there is some delay, and therefore you hear echo. So start reading the advice that other people are offering. One thing that other people haven't mentioned, is that if you are not in the US, then you should also set the OPERMODE value to your country. Interesting, If I call someone who doesn't pick up right away, I can still hear myself echo really badly if I talk into the phone while it is still ringing at the other end. Does this help?? Strange/interesting, but I personally don't know enough about this to comment further. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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 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] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Have you patched the openh323 code with the file included in asteris-oh323-0.7.1 ? Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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] IPSwitchBoard is now Event Driven
Version 0.85 - 13. April 2005. * IPSwitchBoard is now event-driven - much less load on server * Major bug fixes. FREE Download here: http://ipswitchboard.thorben.dk IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: * Unattended/attended transfers. * Park calls and retrieve/forward them again. * Organize all your SIP and IAX extensions (automatically retrieved from Asterisk). * Monitor all extensions. * Monitor all queues. * Monitor Agents. * Monitor Parked Calls. * Dynamically log extensions in and out of queues. * Integration with CRM software on the web. * Drop any active call. * Import/Export extensions to/from Asterisk Server DB. * Set Do Not Disturb on Extensions and give a reason. * Speed Dialling. ___ 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] binding Asterisk to virtual IP
On 4/12/05, Xu Wang [EMAIL PROTECTED] wrote: Our Asterisk works fine with 'real' IP. But when we change the domain to a virtual IP, the audio stream probably goes to the 'real' IP. There is no sound coming back. Asterisk log shows that it does not hang up. Do you know what might be wrong? This sounds like the bug currently being worked on in CVS. Please test the patch and submit feedback to the bug tracker. To quote the bug description: Currently if we have Asterisk SIP channel driver binding to all interfaces, and eth0 has many subnets attached to it (a primary 10.1.200.1, and then alias interfaces eth0:1 with 10.1.201.1, eth0:2 with 10.1.202.1, eth0:3 with 10.1.203.1.. If an INVITE is sent to Asterisk on 10.1.202.1 (eth0:2) the response is always returned to 10.1.200.1. We need it to come back to 10.1.202.1. Here is a direct link to the bug and much more information. http://bugs.digium.com/bug_view_page.php?bug_id=0002358 Thanks, Leif Madsen http://www.leifmadsen.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] i need help
Hello i uses TDM11B, and i successfully use exten = _.,1,Dial(SIP/[EMAIL PROTECTED],10) to make a channel with another PC have the same subnet and gateway . the problem comes when i try to dial another PC have defferent subnet and gateway. it gives the following message: WARNING[14852]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time please guide me to solve this problem thanks in advance Reserve your free [EMAIL PROTECTED], http://www.egypt.com Spam free Virus clean web based mail service ___ 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] Version 0.80 of IPS released
| I removed the old version, deleted the install directory, and installed a | new version, then changed the config so it connects to a different pbx. I | am | still seeing all the old extensions, nothing from the new pbx, even though | ipswitchboard is connecting to the new pbx. Where are the extensions being | cached and how do I do a completely new install? Can you add the ability | to | switch pbx's? | | New PBX: | == Manager 'anguilla' logged off from 206.48.59.5 | == Parsing '/etc/asterisk/manager.conf': Found | == Manager 'anguilla' logged on from 206.48.59.5 | | Chris Mason | www.anguillaguide.com The only file you need to delete is ...Documents\IPSwitchBoard\config.xml That file contains your configuration. I will look at switching servers. thorben ___ 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] IAX2 - Between two ASterisk Servers
Ok, it seems to be working to some degree. The IAX debug comes back with No such context/extension I did a search on the archive and the only thing I could find is that the receiving machine needs the context= I have this in the user section of the IAX.conf. It points to the same context the zap channels use. I am trying to dial a SIP extension.Any ideas? Regards, Chris - Original Message - From: Colin Anderson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 4:40 PM Subject: RE: [Asterisk-Users] IAX2 - Between two ASterisk Servers The AMP configuration didn't work so I decided to work up from the Wiki example. Can anyone help? This is a good config using AMP: http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515 HTH ___ 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 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] DS3000P - 20 E1 capacity on single card
Andrew Kohlsmith wrote: Yes, but then what are you doing with it? You're shuttling the new data to/from a network card in a lot of cases. Combined with other traffic over the PCI bus for normal system operation I could see you coming close to the limitations of regular ole PCI. Absolutely. The DS3000P will definitely support PCI-X, and probably bus speeds of 100MHz or higher, so at least if your system has that you will have plenty of bus capacity. Many servers nowadays actually have their NICs on a separate PCI bus as well, so the TDM and NIC cards won't be contending for the same resources. True enough, but you still need to marshall the data going between PCI busses and to system memory. Certainly not impossible problems to overcome but they do add to the fun of getting a low latency VOIP system together. Very true; realistically, modern PC hardware has more than enough bandwidth to do what is required. The real issue is timing, based on contention for resources, and how that impacts latency. The existing boxes out there (not PCs) that handle DS3 have far lower performance metrics than a 3GHz P4 or similar system :-) ___ 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] DS3000P - 20 E1 capacity on single card
Steve Underwood wrote: Since encoding typically requires 5 times as much compute as decoding, for CELP based codecs, an encode onyl board would not be as dumb as it seems at first sight :-) Hah! I knew someone would say that! ___ 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] Acceptable voice time delay
--- Sean Kennedy [EMAIL PROTECTED] wrote: chawki, That additional 2.5 seconds of delay may be any combination of things, but I would look first to your ISP and their backbone. I will try new isp in two weeks with better routing. Meanwhile, is there any known latency issues if Asterisk is behind a nat vs. asterisk with a real ip. this is how i configured my voipjet contextin iax.conf file [voipjet] type=peer host= 216.118.117.46 secret= secret number auth=md5 notransfer=yes context=default nat=yes careinvite=no I have just added nat=yes and careinvite=no thinking this might be the cause of the delay, but no change does iax maintains open communication at all times once the call starts? Good luck! Sean chawki hammoud wrote: thank you Rob: the problem is that I am experiencing about 3 secs latency although the ping is 600ms which is a round trip packet travel time. so i should experience about half a sec latency including the voipjet server response and the latency to the pstn. that is annoying, but nothing compared to about 3 sec. do you think the rest of the delay is due to voipjet slow response to the pstn network or some other issues would you be bale to clculate where the 3 sec is comming from thanks. Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. ___ 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 __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ 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] i need help
If you have two devices on the same subnet and both are registered to *, then calls will complete. If the devices are on separate subnets, then you have to address issues such as... Firewalling? Using NAT? Routing in general? SIP won't natively traverse firewalls so that would be a starting point... Search the Wiki www.voip-info.org and Google using: site:lists.digium.com some paramater Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 13, 2005 7:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] i need help Hello i uses TDM11B, and i successfully use exten = _.,1,Dial(SIP/[EMAIL PROTECTED],10) to make a channel with another PC have the same subnet and gateway . the problem comes when i try to dial another PC have defferent subnet and gateway. it gives the following message: WARNING[14852]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) == No one is available to answer at this time please guide me to solve this problem thanks in advance Reserve your free [EMAIL PROTECTED], http://www.egypt.com Spam free Virus clean web based mail service ___ 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 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] SIP registration fails
Title: SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip=myexternaldyndnsname realm=myrealm context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register = mysipid:mysippass@sip.web.de/mysipid [webde] type=friend username=mysipid secret=mysippass host=sip.web.de fromuser=mysipid fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between register= and [webde] done? many thanks. ___ 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] Acceptable voice time delay
--- Sean Kennedy [EMAIL PROTECTED] wrote: chawki, That additional 2.5 seconds of delay may be any combination of things, but I would look first to your ISP and their backbone. I will try new isp in two weeks with better routing. Meanwhile, is there any known latency issues if Asterisk is behind a nat vs. asterisk with a real ip. this is how i configured my voipjet contextin iax.conf file [voipjet] type=peer host= 216.118.117.46 secret= secret number auth=md5 notransfer=yes context=default nat=yes careinvite=no I have just added nat=yes and careinvite=no thinking this might be the cause of the delay, but no change does iax maintains open communication at all times once the call starts? thanks Good luck! Sean chawki hammoud wrote: thank you Rob: the problem is that I am experiencing about 3 secs latency although the ping is 600ms which is a round trip packet travel time. so i should experience about half a sec latency including the voipjet server response and the latency to the pstn. that is annoying, but nothing compared to about 3 sec. do you think the rest of the delay is due to voipjet slow response to the pstn network or some other issues would you be bale to clculate where the 3 sec is comming from thanks. Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. ___ 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 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] FRAME_CONTROL (5) dropping calls on PRI
After turning on full debug logs and getting users to report dropped calls, I have had 2 dropped calls in as many days. drop1: Apr 11 16:11:14 DEBUG[15029]: Got a FRAME_CONTROL (5) frame on channel Zap/3-1 drop2:Apr 13 09:13:37 DEBUG[5563]: Got a FRAME_CONTROL (5) frame on channel Zap/1-1 I can post full debug logs if anyone thinks that would help. Per yesterday's Line Noise thread, I have used zttest to check timing and used hdparm to set hda to udma2. zttest mostly reports 100% to 99.9875% with a few times dipping to 99.90%. I'm modifying zttest to log the time (like the log files) and to only report when under 99.98% (I will post a patch and bug when it is ready) so I can run it for a day without logs every second. Is anyone else seeing this and or have a solution (other than to correlate timing drops with this FRAME_CONTROL (5) )? I will post back when I have more info. Jeb Campbell [EMAIL PROTECTED] ___ 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] IAX2 - Between two ASterisk Servers
Use the same context that your SIP phones on the target Asterisk server use. If you are using AMP, try the context from-internal -Original Message- From: Chris [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 13, 2005 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 - Between two ASterisk Servers Ok, it seems to be working to some degree. The IAX debug comes back with No such context/extension I did a search on the archive and the only thing I could find is that the receiving machine needs the context= I have this in the user section of the IAX.conf. It points to the same context the zap channels use. I am trying to dial a SIP extension.Any ideas? Regards, Chris - Original Message - From: Colin Anderson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 4:40 PM Subject: RE: [Asterisk-Users] IAX2 - Between two ASterisk Servers The AMP configuration didn't work so I decided to work up from the Wiki example. Can anyone help? This is a good config using AMP: http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515 HTH ___ 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 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] 3-Way Calling in Asterisk
Anybody doing it with Grandstream handytone ATA 286? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven Sent: Miércoles, 13 de Abril de 2005 08:29 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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 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] snom and hint priority
(boy mail in this list piles up fast when I can't check it) On Apr 8, 2005, at 10:03 AM, Michael George wrote: - It appears that the extension used with the hint must be the same as the extension used to dial that channel. So if extension 22 will ring Zap/2, then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will not. Why is that? The extension is how asterisk maps SIP URLs to chunks of your dialplan -- if you program a button on a snom to dest sip:[EMAIL PROTECTED], the phone will use that same URL for both dialing and subscribing to extension state. Unless you have a phone that lets you specify different URLs for dialing and subscribing to state, they have to match in asterisk. - If I am correct in the above, then there is no way for me to monitor a channel that is not an extension. As an example, I have a TDM400 with 3 FXS (Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel for dialing out. I can monitor the states of the extensions with extension entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4. Indeed, Zap/4 has no extension. Is it not possible to monitor that line, then? There has to be a SIP URL for the phone to subscribe to -- if you put: exten = zap4,hint,Zap/4 in your extensions.conf (with no zap4,1,... entry) it wouldn't be dialable (although the phone would still try if you pushed it) but would have a valid SIP URL. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ 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] ZAP channel hangs up with no apparent reason
Hi, Recently I've been having strange behaviour on my calls to PSTN, when dialing from any extension to the PSTN through ZAP the line hangs up after exactly 3:03 mins., tried to look everywhere for a string defining this timing but of no use, I even set the AbsoluteTimeout in the dialplan to 0 but still the problem persisted, any suggestions? Ezabi signature.asc Description: OpenPGP digital signature ___ 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] Newbie Question on how to handle main office number
Hi we are looking to swap out an old version of Cisco Call Manager for asterisk and are trying to work out the best way to handle the main office number. We will have about 35 phones and we have PRI from Colt, we are london based. At present when a call is made to the main number and is not answered or is engaged we forward the call to a shared number that every phone in the office is registered to and so can answer. But this limits us to only receiving two calls at once to our main number. We like the way this works apart from the limit of two calls at once and are trying to work out the best way to implement this in asterisk. Although we have looked at queue's we don't want everybody to having login every morning ?. Would a solution where we forward calls from the main number to an extension that calls all the phones, be a solution ? exten = 2000,1,Dail(SIP/101,SIP/102,SIP/103 etc) Or would this get engaged after the first call is answered on it, if could I have a number of these that roll over to each other ? Or are there other ways to handle the main number ? Thanks for any help Nick ___ 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] Local Echo
On Tue, Apr 12, 2005 at 05:43:18PM -0700, Noah Silverman wrote: Great suggestion. I'll try it ASAP. Where do I get fxotune? It's in CVS-HEAD zaptel. You'lll need to use the CVS-HEAD zaptel drivers as well, since there is a new IOCTL for doing echo tuning. Matthew Fredrickson ___ 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]Unable to register license for G729 codec
Hi, I bought the license for codec g.729a from digium and am now facing some problem registering the codec with them. i got the following message. -- ./register G729-**key** Digium Product Registration Copyright (C) 2004, Digium, Inc. Analyzing key 'G729-**key**' Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! - Kindly give ur valuable suggestion. Thanks, Firdosh ___ 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] Local Echo
On Tue, Apr 12, 2005 at 07:03:26PM -0700, Bashir Ullah - www.Lamsre.Com wrote: hi i did not find fxotune under zapte-1.0.6 , please let me know is it different module , need to install seperate, please show me the way , i am having same echo problem and finding its solution for mt tdm fxo. It's in CVS-HEAD zaptel. You'll need to use the CVS-HEAD drivers as well for it to work. Matthew Fredrickson ___ 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] CVS-HEAD Zaptel with 1.0.x CVS Asterisk
Digium support suggested today that I run CVS-HEAD zaptel with 1.0.x CVS Asterisk. This seems totally wrong to me. Can others confirm? --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] DS3000P - 20 E1 capacity on single card
On April 13, 2005 10:57 am, Kevin P. Fleming wrote: Very true; realistically, modern PC hardware has more than enough bandwidth to do what is required. The real issue is timing, based on contention for resources, and how that impacts latency. The existing boxes out there (not PCs) that handle DS3 have far lower performance metrics than a 3GHz P4 or similar system :-) Well yes, but they're not a general computing platform either and their I/O design is quite different. They could spank any PC in terms of concurrent I/O without even breaking a sweat. :-) -A. ___ 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] Zaptel and Fritz Card
Hi Oliver, I am trying to install only the Fritz Card. But according to the instructions on: http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install it doesnt work. The directories, even the changes that they suggest on the makefile are not there!! I am really disappointed I have been on this for hours!! ___ 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] PRI Errors with TE110P
Aaron Mathews wrote: I'm having a problem with a new digium te110p card. I'm running it on a T1 with PRI signalling, and everything works fine *except* I get errors every few minutes that look like the following: Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on 40 failed: Unknown error 500 Apr 11 23:23:04 NOTICE[10251]: chan_zap.c:6708 pri_dchannel: PRI got event: 8 on span 1 Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on 40 failed: Unknown error 500 Apr 11 23:23:04 NOTICE[10251]: chan_zap.c:6708 pri_dchannel: PRI got event: 6 on span 1 And continue on and on just like that. I found some old mailing lists posts from the beginning of 2004 that seemed to indicate that this was a 'frame buffering' problem, and that digium was working on a fix- is this still the case? Is there a fix? Something is locking interrupts on your system for so long that the Digium card is losing data from the PRI. It could just be a crappy motherboard (the SuperMicro board I got recently did this). Usually it's caused by the IDE and you can use the various things listed in the mailing list archives like unmasking interrupts, enabling DMA, etc to reduce the time interrupts are locked. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] 3-Way Calling in Asterisk
As far as I can see, never gonna happen with an ATA. ATA is your end point and has no exploitable features like that. It just connects your analog phone to a digital network. Meetme or Conference are probably your only bet in that case... http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, April 13, 2005 8:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 3-Way Calling in Asterisk Anybody doing it with Grandstream handytone ATA 286? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven Sent: Miércoles, 13 de Abril de 2005 08:29 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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 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 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] 3-Way Calling in Asterisk
I do it all the time.. Just like a standard phone Call someone, flash hook, get second dial tone, call another person, flash hook and all three are connected.. I didn't have to do anything, this works fine.. The one caveat to this is I cannot get it to work on my analog line (Don't know how to send the zaptel driver a flash hook event), so it only works if I use my VOIP provider.. - Andre -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Allen Niven Sent: Wednesday, April 13, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk i do it on the 79xx, the polycom series and sipura 841 just on on the fone display aram wrote: Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ 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 -- Allen Niven GlobalFone 350 Fifth Avenue #6206 Empire State Building New York, NY 10118 +1-212-678-4381 office +1-646-246-7415 cell http://www.GlobalFone.biz Instant Messaging Accounts ICQ 137763656 Yahoo Messenger [EMAIL PROTECTED] MSN Messenger [EMAIL PROTECTED] PLEASE NOTE I NEVER NEVER NEVER RECEIVE EMAILS ON HOTMAIL OR YAHOO ___ 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 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]Unable to register license for G729 codec
Contact Digium For this issue -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed Firdosh Nasim Sent: mercredi 13 avril 2005 16:12 To: asterisk-users@lists.digium.com Cc: Mohammed Firdosh Nasim Subject: [Asterisk-Users]Unable to register license for G729 codec Hi, I bought the license for codec g.729a from digium and am now facing some problem registering the codec with them. i got the following message. -- ./register G729-**key** Digium Product Registration Copyright (C) 2004, Digium, Inc. Analyzing key 'G729-**key**' Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! - Kindly give ur valuable suggestion. Thanks, Firdosh ___ 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 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]Unable to register license for G729 codec
On Wed, 13 Apr 2005, Mohammed Firdosh Nasim wrote: Hi, I bought the license for codec g.729a from digium and am now facing some problem registering the codec with them. i got the following message. Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! Perhaps you have a firewall/packet filter that's stopping you from connected to Digium's key server? It's working from here. $ telnet 216.207.245.3 5646 Trying 216.207.245.3... Connected to 216.207.245.3. Escape character is '^]'. 220 Welcome to cpsignd -- Jon Lewis | I route Senior Network Engineer | therefore you are Atlantic Net| _ http://www.lewis.org/~jlewis/pgp for PGP public key_ ___ 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] x-ten lite error
Title: Re: [Asterisk-Users] x-ten lite error Make sure the codec's are all highlighted: This is a common error. Robert Andrew Keller Ferndale School District #502 [EMAIL PROTECTED] 360-383-9228 PH. 360-383-9218 FAX Paving the way for tomorrows genius. From: [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 13 Apr 2005 14:51:56 +0200 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] x-ten lite error Hallo, I have just set up [EMAIL PROTECTED] and configured two extensions, 200 and 201 with x-ten lite. Ext 201 seems to be ok (i tried to dial 200 and the system answered that the extension was not available atg the moment and let me leave a voicemail message), Ext 200 seems to be blocked: for any number that i try to dial I always receive the busy tone and the message Call failed: 403 forbidden appear on the softphone screen. Any suggestion? tia brgs Francesco ___ 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 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] Pretty Voicemail Docs
Has anyone written up pretty voicemail user docs? I think voicemail is so easy even my cat can use it. However, my users are complaining about lack of docs for voicemail. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
Yes, I followed the instructions at: http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en Guillermo Salas M wrote: On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Have you patched the openh323 code with the file included in asteris-oh323-0.7.1 ? Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Newbie Question on how to handle main office number
Check out AMP to see how call groups are used. http://www.voip-info.org/wiki-Asterisk+Management+Portal You group your phones, available handsets ring. You can roll from group to group however you want. Just a matter of writing the correct dialplan. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Teagle Sent: Wednesday, April 13, 2005 8:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newbie Question on how to handle main office number Hi we are looking to swap out an old version of Cisco Call Manager for asterisk and are trying to work out the best way to handle the main office number. We will have about 35 phones and we have PRI from Colt, we are london based. At present when a call is made to the main number and is not answered or is engaged we forward the call to a shared number that every phone in the office is registered to and so can answer. But this limits us to only receiving two calls at once to our main number. We like the way this works apart from the limit of two calls at once and are trying to work out the best way to implement this in asterisk. Although we have looked at queue's we don't want everybody to having login every morning ?. Would a solution where we forward calls from the main number to an extension that calls all the phones, be a solution ? exten = 2000,1,Dail(SIP/101,SIP/102,SIP/103 etc) Or would this get engaged after the first call is answered on it, if could I have a number of these that roll over to each other ? Or are there other ways to handle the main number ? Thanks for any help Nick ___ 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 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] VAD/DTX implementation through zaptel cards
No. parijat wrote: Pls could u be more elaborate as I am new to asterisk.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, April 13, 2005 7:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards The only time PLC makes sense is thwn you are converting FROM VoIP to something else. So PLC would be done on chan_sip or chan_IAX, or chan_h323 on the receiving end. This is for 1.0.x. For CVS-HEAD you would want to do this on the receiving side in the PLC stuff. parijat wrote: Hi, Thanks for helping me out. I want to clear out few more points 1) zaptel cards receive PCM from PSTN. In what form do they give it to asterisk. Do Zaptel cards CODE/DECODE PCM from PSTN to RTP or zaptel cards forward PCM to asterisk which converts it to RTP. 2) If asterisk does that conversion then, using which file does it convert. I want to change code of that file so that I can implement VAD. 3) If all this is not possible then why they have give so many codec files in asterisk. Regards, Parijat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 12, 2005 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards Steve Kann wrote: Eric Wieling wrote: [EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. Doing VAD on audio coming _from_ the TDM world certainly is something you might want to do, to dramatically reduce the bandwidth you consume when sending the audio via VoIP channels. This kind of thing is not presently implemented in *, though, but it could be. (note: doing it well will require a bunch of CPU, though. I wonder if it could be done in the same DSP that is doing echo-cancellation on the new TE4xxP boards? Unless Digium's plans changed since the last time I spoke to Mark, the answer would be no. I believe they are using a dedicated function echo canceller device. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Newbie Question on how to handle main office number
That is really the beauty of a good IVR menu design. In a good design not only do you eliminate the everyone can answer every call it also benefits the caller because they get directed to the person/dept they need to get to faster and it solves the one call at a time problem. A good IVR design does not have to be complicated, a very basic one can be If you know your party's extension, please dial it now Press 1 for sales Press 2 for support Press 3 for marketing Press 0 for the operator (ring all phones) Press # for company directory Then have each group in a ring group. A simple routing of calls like that will save everybody time improve call effeciency. Kerry Garrison http://www.geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Teagle Sent: Wednesday, April 13, 2005 8:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newbie Question on how to handle main office number Hi we are looking to swap out an old version of Cisco Call Manager for asterisk and are trying to work out the best way to handle the main office number. We will have about 35 phones and we have PRI from Colt, we are london based. At present when a call is made to the main number and is not answered or is engaged we forward the call to a shared number that every phone in the office is registered to and so can answer. But this limits us to only receiving two calls at once to our main number. We like the way this works apart from the limit of two calls at once and are trying to work out the best way to implement this in asterisk. Although we have looked at queue's we don't want everybody to having login every morning ?. Would a solution where we forward calls from the main number to an extension that calls all the phones, be a solution ? exten = 2000,1,Dail(SIP/101,SIP/102,SIP/103 etc) Or would this get engaged after the first call is answered on it, if could I have a number of these that roll over to each other ? Or are there other ways to handle the main number ? Thanks for any help Nick ___ 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 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] ZAP channel hangs up with no apparent reason
On April 13, 2005 11:20 am, Ezabi wrote: Recently I've been having strange behaviour on my calls to PSTN, when dialing from any extension to the PSTN through ZAP the line hangs up after exactly 3:03 mins., tried to look everywhere for a string defining this timing but of no use, I even set the AbsoluteTimeout in the dialplan to 0 but still the problem persisted, any suggestions? Are you using busydetect or callprogress in zapata.conf? -A. ___ 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] DS3000P - 20 E1 capacity on single card
Kevin P. Fleming wrote: Andrew Kohlsmith wrote: Yes, but then what are you doing with it? You're shuttling the new data to/from a network card in a lot of cases. Combined with other traffic over the PCI bus for normal system operation I could see you coming close to the limitations of regular ole PCI. Absolutely. The DS3000P will definitely support PCI-X, and probably bus speeds of 100MHz or higher, so at least if your system has that you will have plenty of bus capacity. Many servers nowadays actually have their NICs on a separate PCI bus as well, so the TDM and NIC cards won't be contending for the same resources. True enough, but you still need to marshall the data going between PCI busses and to system memory. Certainly not impossible problems to overcome but they do add to the fun of getting a low latency VOIP system together. Very true; realistically, modern PC hardware has more than enough bandwidth to do what is required. The real issue is timing, based on contention for resources, and how that impacts latency. The existing boxes out there (not PCs) that handle DS3 have far lower performance metrics than a 3GHz P4 or similar system :-) That is a meaningless comparison. Those boxes don't the audio touch the processor, or its buses. Regards, Steve ___ 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] Busy line status and chan_capi?
Hi Kib, What exactly is it that you want to do? If you have a direct dial-in (DDI) number that goes to a certain extension then you can handle this pretty easily in your dial plan - check out this snippet below from one of our customers' machines. This example is pretty basic, but it works fine for their requirements: exten = 290,1,Answer exten = 290,2,Dial(SIP/9290,,t,) exten = 290,3,Voicemail(9290) exten = 290,4,Congestion The extension (290 in this example) is the DDI (known as a DID) number that the telco presents on the line (this customer has 10 DDI's) and that CAPI presents when the line rings. This is used to decide which extension to route the call to. By default SIP will use call waiting, which this particular customer doesn't like, so they dial *71 from each extension to cancel it. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Kib Eki wrote: What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Kib ___ 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 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] SNOM 220 with 7 lines
It seems like you missed the difference between a line configuration and the configuration of the programable buttons. In first they are not related to each other in any way. So configure your seven lines on the web pages and then you can configure your 25 buttons. There is (currently) no way to setup more then 7 lines on a snom phone, no matter how many sidecars you have. Sorry for that. Regards Nils On Wednesday 13 April 2005 16:06, Michael Welter wrote: I have a SNOM 220 with a 20-button sidecar. The configuration for the five lines (buttons) on the main phone is straigth forward: display name, account, password, registrar. I would like to get each of the sidecar buttons to register with Asterisk in Line mode so that I can have incoming calls from each DID number appear on a separate button. For the sidecar buttons, however, all I have is a URL. I've tried sip:[EMAIL PROTECTED] which won't work because there's no password. I've tried sip:421:[EMAIL PROTECTED], but the : is converted into a %3a and it doesn't register. \: doesn't work either. Does anyone have any insight? Thanks, ___ 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 -- snom technology AGPascalstrasse 10bD-10581 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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
