[Asterisk-Users] Build a complex IVR?
Hi,every one! I would like to know if Asterisk could be used to build a IVR with complex flow? To provide a complex sample would be appreciated. Regards. john
Re: [Asterisk-Users] ATA-186 and fake ring
On 2003-03-12 at 09:44, you wrote: > > Who is generating this ringback? The ATA or asterisk? > Find out by doing a trace. If you're using callprogress, then you should > see a 180 Ringing sent to the ATA when we detect ringing on the FXO. If > you're not using call progress, then we should not be sending 180 ringing. We're not using callprogress (at least it's not set in zapata.conf). I also tried explicitly setting it to 'no', reloading, and trying again. No change. I do get a 180 Ringing. I am dialing from my ATA-186 to 18189950699, a telco busy test. Yet all I hear is a ring, though on one call I heard a short blip of busy before the ring started. [See attachment for the trace output] > You can also use the new "debug channel Zap/1-1" to see if the FXO is > ringing. If I do a 'show channel' on it, I get: Zap/21-1 (intrunk6197474525 1 ) Ringing AppDial (Outgoing Line) SIP/0054-b2bc (outtrunk 8189950699 2 )Ring Dial Tor/g1/BYEXTENSION That 6197474525 is strange. That's probably the DNIS used on the last incoming call to that channel, but has nothing to do with my outgoing call. I was calling from 6193640054 (SIP 0054). My definition in sip.conf is: [0054] type=friend insecure=yes secret=myownsecret callerid="Jim Gottlieb <(619) 364-0054>" ; dynamic binding seems to time-out; try defaultip host=dynamic defaultip=192.168.40.90 ; need to set the following so we can use voicemail and other DTMF apps dtmfmode=rfc2833 [I dialed...] Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: User-Agent: Cisco ATA v2.15 ata18x (020927a) Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5680 5680 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Interface is eth0 IP Address is 198.51.175.9 Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: ;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="028f4554" Content-Length: 0 to 192.168.40.90:5060 Sip read: > ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: ;tag=06e4b177 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="0054",realm="asterisk",nonce="028f4554",ur i="sip:[EMAIL PROTECTED]",response="a8dbe8f8d6faee139756514c82cad48f" Expires: 300 Content-Length: 247 Content-Type: application/sdp v=0 o=0054 5686 5686 IN IP4 192.168.40.90 s=ATA186 Call c=IN IP4 192.168.40.90 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.40.90 : 5060 (non-NAT) Capabilities: us - 14, them - 13, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 18189950699 in sip Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: ;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.168.40.90:5060 We're at 198.51.175.9 port 53614 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: ;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 5860 5860 IN IP4 198.51.175.9 s=session c=IN IP4 198.51.175.9 t=0 0 m=audio 53614 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 TELEPHONE-EVENT/8000 a=fmtp:101 0-16 to 192.168.40.90:5060 Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.40.90:5060 From: ;tag=850095511 To: ;tag=5b1ade90 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Contact: Content-Length: 0 to 192.168.40.90:5060 [At this pont I was hearing ringback tone. Then I hung up...] Sip read: > CANCEL sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 19
Re: [Asterisk-Users] Help With Music On Hold
Did you remember to move to mpg123 and not use mpg321 which is often
aliased to mpg123?
Mark
On Wed, 12 Mar 2003, Walt Davis wrote:
> It used to work just fine and I can only think of three possibilities:
>
> 1) Moved asterisk onto a different server and that server does not have a
> sound card?
>
> 2) Rebuilt and hardened asterisk on a Mandrake 9.0 distro, so perhaps I
> left out some modules or something.
>
> 3) Upgraded from the developer kit light to a single span T1.
>
> Can anybody confirm or deny these as possibilities?
>
> > Asterisk -rvvv
> > -- Starting simple switch on 'Zap/7-1'
> > -- Executing MusicOnHold("Zap/7-1", "") in new stack
> > -- Started music on hold, class 'default', on Zap/7-1
> > -- Stopped music on hold on Zap/7-1
> > == Spawn extension (home, 6, 1) exited non-zero on 'Zap/7-1'
> > -- Hungup 'Zap/7-1'
> >
> > As you can see MusicOnHold is starting and stopping but I dont hear
> > anything, just dead silence.
> >
> > My music on hold config:
> > default => mp3:/var/lib/asterisk/mohmp3, -z.
> >
> > All MP3 Files in this dirctory have global read access.
> >
> > I'm getting a lot of these in the debug log:
> > Mar 11 14:12:10 DEBUG[3076]: File res_musiconhold.c, Line 240
> > (monmp3thread): Read 100 bytes of audio while expecting 640
> >
> > and these in the Message log:
> > Mar 11 14:10:24 WARNING[1024]: File chan_oss.c, Line 419
> > (soundcard_init)
> >
> > Any ideas?
> >
> > --
> > Walt Davis
> > www.waltdavis.net
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ___
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>
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Re: [Asterisk-Users] Music - Hold -> live sound source ?
working examples would be appreciated ;-) On Wed, 12 Mar 2003 23:32:05 -0600 (CST), Mark Spencer wrote: >In principle you could use a named pipe in the filesystem right? Then you >don't have to fork/exec anything (man mkfifo) > >Mark > >On Wed, 12 Mar 2003, Chris Albertson wrote: > >> >> Try this. Make a "pipe" called "live.mp3" then use the normal >> Asterisk music on hold function for play the pipe. >> >> Next you will need a very simple copy type script to read >> dev/audio filtr it through an MP3 encoder and write it to the >> pipe. >> >> Somehow you fork/exec the script just before connecting the >> call to music on hold. >> >> >> --- Steven Critchfield <[EMAIL PROTECTED]> wrote: >> > On Wed, 2003-03-12 at 18:35, Gary wrote: >> > > Hi folks, >> > > >> > > I am looking for a way to actually have a live feed for a music on >> > hold >> > > channel. >> > > >> > > Basically this is for a local radio station feed so people can ring >> > to >> > > get the lastest report... Sort of like the US vhf weather >> > channels >> > > >> > > ANy ideas please ?? >> > >> > If you are only wanting to use 1 line, then you could use chan_oss >> > and >> > autoanswer to connect an incoming call to the soundcard. If you want >> > more, you may need to write an app that would access the soundcard or >> > an >> > app that could multiplex the soundcards input. >> > -- >> > Steven Critchfield <[EMAIL PROTECTED]> >> > >> > ___ >> > Asterisk-Users mailing list >> > [EMAIL PROTECTED] >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> = >> Chris Albertson >> Home: 310-376-1029 [EMAIL PROTECTED] >> Cell: 310-990-7550 >> Office: 310-336-5189 [EMAIL PROTECTED] >> KG6OMK >> >> __ >> Do you Yahoo!? >> Yahoo! Web Hosting - establish your business online >> http://webhosting.yahoo.com >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help With Music On Hold
It used to work just fine and I can only think of three possibilities:
1) Moved asterisk onto a different server and that server does not have a
sound card?
2) Rebuilt and hardened asterisk on a Mandrake 9.0 distro, so perhaps I
left out some modules or something.
3) Upgraded from the developer kit light to a single span T1.
Can anybody confirm or deny these as possibilities?
> Asterisk -rvvv
> -- Starting simple switch on 'Zap/7-1'
> -- Executing MusicOnHold("Zap/7-1", "") in new stack
> -- Started music on hold, class 'default', on Zap/7-1
> -- Stopped music on hold on Zap/7-1
> == Spawn extension (home, 6, 1) exited non-zero on 'Zap/7-1'
> -- Hungup 'Zap/7-1'
>
> As you can see MusicOnHold is starting and stopping but I dont hear
> anything, just dead silence.
>
> My music on hold config:
> default => mp3:/var/lib/asterisk/mohmp3, -z.
>
> All MP3 Files in this dirctory have global read access.
>
> I'm getting a lot of these in the debug log:
> Mar 11 14:12:10 DEBUG[3076]: File res_musiconhold.c, Line 240
> (monmp3thread): Read 100 bytes of audio while expecting 640
>
> and these in the Message log:
> Mar 11 14:10:24 WARNING[1024]: File chan_oss.c, Line 419
> (soundcard_init)
>
> Any ideas?
>
> --
> Walt Davis
> www.waltdavis.net
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Re: [Asterisk-Users] SIP and MWI 7960
> Is anyone using MWI on 7960's when Asterisk is ONLY being used > for voicemail, and not for a gatekeeper? > > If anyone has MWI working successfully with 7960's, would it > be possible to get a dump of a successful NOTIFY message that > turns a light on/off? Just put "mailbox=1234" in your sip friend/peer for the phone. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music - Hold -> live sound source ?
In principle you could use a named pipe in the filesystem right? Then you don't have to fork/exec anything (man mkfifo) Mark On Wed, 12 Mar 2003, Chris Albertson wrote: > > Try this. Make a "pipe" called "live.mp3" then use the normal > Asterisk music on hold function for play the pipe. > > Next you will need a very simple copy type script to read > dev/audio filtr it through an MP3 encoder and write it to the > pipe. > > Somehow you fork/exec the script just before connecting the > call to music on hold. > > > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > On Wed, 2003-03-12 at 18:35, Gary wrote: > > > Hi folks, > > > > > > I am looking for a way to actually have a live feed for a music on > > hold > > > channel. > > > > > > Basically this is for a local radio station feed so people can ring > > to > > > get the lastest report... Sort of like the US vhf weather > > channels > > > > > > ANy ideas please ?? > > > > If you are only wanting to use 1 line, then you could use chan_oss > > and > > autoanswer to connect an incoming call to the soundcard. If you want > > more, you may need to write an app that would access the soundcard or > > an > > app that could multiplex the soundcards input. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > = > Chris Albertson > Home: 310-376-1029 [EMAIL PROTECTED] > Cell: 310-990-7550 > Office: 310-336-5189 [EMAIL PROTECTED] > KG6OMK > > __ > Do you Yahoo!? > Yahoo! Web Hosting - establish your business online > http://webhosting.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music - Hold -> live sound source ?
Try this. Make a "pipe" called "live.mp3" then use the normal Asterisk music on hold function for play the pipe. Next you will need a very simple copy type script to read dev/audio filtr it through an MP3 encoder and write it to the pipe. Somehow you fork/exec the script just before connecting the call to music on hold. --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Wed, 2003-03-12 at 18:35, Gary wrote: > > Hi folks, > > > > I am looking for a way to actually have a live feed for a music on > hold > > channel. > > > > Basically this is for a local radio station feed so people can ring > to > > get the lastest report... Sort of like the US vhf weather > channels > > > > ANy ideas please ?? > > If you are only wanting to use 1 line, then you could use chan_oss > and > autoanswer to connect an incoming call to the soundcard. If you want > more, you may need to write an app that would access the soundcard or > an > app that could multiplex the soundcards input. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Web Hosting - establish your business online http://webhosting.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect & caller ID
The friend would only happen if the "From: " was iconnect. Unfortuantely SIP does not differentiate a user from Caller*ID. The only way to make the peer match would be if we matched the peer based on IP address. Mark On Wed, 12 Mar 2003, Jim Archer wrote: > Hi All... > > We have found that the caller ID information presented to some one we call > from Asterisk using iconenct is not predictable. The caller ID will be > unavailable or else deltathree. I have in sip.conf: > > [iconnect] > type=friend > username=41306756 > password=2264 > host=natrelay.deltathree.com > callerid="My COmpany, Inc." <1 401 nnn > > txgain = 5.0; > rxgain = 5.0; > > Has anyone else seen this problem? > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music - Hold -> live sound source ?
> it does need to service multiple lines at the same time. > > Also as a background music (on loudspeaker type phones) would be nice. > > We are currently playing with festival, so the caller can select which > area they want details for and we hourly download and massage the > hourly updates avail from our weather mob... The music on hold engine was designed to support plugins for this sort of purpose. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
What mode are you running the Phone in? SIP, MCGP, or SCCP (Skinny) You mentioned Call Manager so I will assume SCCP. If that is the case I do not know. However if you are running it in SIP, All you have to do is set # XML URLs services_url: ""; URL for external Phone Services directory_url: "" ; URL for external Directory location logo_url: ""; URL for branding logo to be used on phone display These in you configuration and point it to a webserver. The xml format can be found here. http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/bxtml.htm Hope this helps! Stephen On Wed, 2003-03-12 at 18:59, Mike Reiling wrote: > Anyone know if it is possible to load your own XML scripts on to the > phone, bypassing the Cisco CallManager? I am still waiting for my > phone to arrive, but I have been playing with Cisco's phone services > emulator, and that doesn't seem to like anything I pass to it. > > If it is possible, anyone want to share any sample scripts they have. > > --Mike > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
That is basically what I see as well. I do not see any response coming back with the SIP debug. Could it be a problem with the first header line XXX Need to handle Retransmitting XXX: (or is that something generated by asterisk). Just a thought. I am not a protocol expert so I am just curious. I will try to find out what the registration server is in the morning and that may help as well. Bob From: Masakazu Nakano <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP registration Date: Thu, 13 Mar 2003 09:21:21 +0900 version is 'Asterisk CVS-03/11/03-09:57:33' we can regist to wcom in two ways. first. register => masakazu:[EMAIL PROTECTED] * send REGISTER, but no response from wcom. second. quit * and change the way with number. like this. register => 9706052:[EMAIL PROTECTED] and REGISTER again. in this time,get this result following. mack*CLI> Interface is eth0 IP Address is 210.194.204.16 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:166.60.255.41 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f >From: ;tag=2d864abf To: ;tag=2d864abf Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 166.60.255.41:5060 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:166.60.255.41 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f >From: ;tag=2d864abf To: ;tag=2d864abf Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 166.60.255.41:5060 and keep this trying. I sometime meets segfault.maybe that cause... --- Masakazu Nakano On Tue, 11 Mar 2003 22:34:30 -0600 (CST) Mark Spencer <[EMAIL PROTECTED]> wrote: >> **ASTERISK SIP PACKET >> >> XXX Need to handle Retransmitting XXX: >> REGISTER sip:166.60.255.41 SIP/2.0 >> Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924 >> From: ;tag=08e71f4b >> To: ;tag=08e71f4b >> Contact: >> Call-ID: [EMAIL PROTECTED] >> CSeq: 113 REGISTER >> User-Agent: Asterisk PBX >> Expires: 120 >> Event: registration >> (no NAT) to 166.60.255.41:5060 >> >> > >Do we not receive anything back at all? > >Mark > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and MWI 7960
Two issues-- is anyone using Asterisk as a gatekeeper with cisco 7960 phones and cisco gateways? Experiences, thoughts, etc appreciated. If anyone has moved from/to ser to/from Asterisk, I would be interested in hearing experiences... We have been trying to get message waiting indication working on our 7960's without luck. Is anyone using MWI on 7960's when Asterisk is ONLY being used for voicemail, and not for a gatekeeper? If anyone has MWI working successfully with 7960's, would it be possible to get a dump of a successful NOTIFY message that turns a light on/off? thanks bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnect & caller ID
Hi All... We have found that the caller ID information presented to some one we call from Asterisk using iconenct is not predictable. The caller ID will be unavailable or else deltathree. I have in sip.conf: [iconnect] type=friend username=41306756 password=2264 host=natrelay.deltathree.com callerid="My COmpany, Inc." <1 401 nnn > txgain = 5.0; rxgain = 5.0; Has anyone else seen this problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
Hello The option relaxdtmf=yes improved the accuracy of received DTMF digits on cell phones but did not make DTMF detection 100%. Increasing the rxgain to 15 does improve the accuracy almost to 100%, as long as you let the first few milliseconds of the recording play. The problem now is that once a call finds it's way to my handset the received audio is distorted and too loud to carry on a conversation without some discomfort to the listener. I have tried setting the rxgain to a negative db setting for my S100U and that does not seem to have any affect on how loud the signal is (if it worked it still wouldn't do anything for the distortion though). Is this information helpful to anyone? Is there a way I can get more information that would be helpful to someone who can code a solution? [channels] txgain=0 rxgain=15.0 context = default language = en callwaiting = yes callwaitingcallerid = no threewaycalling = yes transfer = yes cancelforward = yes callreturn = no usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = yes immediate = yes relaxdtmf=yes group = 1 ;use with FXO PCI card signalling = fxs_ks channel => 1-3 echocancel = yes echocancelwhenbridged = yes context = local immediate = no group = 2 txgain=15.0 rxgain=-6.0 ;use with FXS USB card signalling = fxo_ks callerid = "Brian Schrock" <(614) 798-9106> mailbox=2244,2245,2246 channel => 4 On Wednesday, March 12, 2003, at 03:46 PM, James Hines wrote: On Wed, 2003-03-12 at 14:32, Martin Pycko wrote: You may try to add relaxdtmf=yes just before channel => 4 in zapata.conf Thanks! This has solved the problem for the test phone! I will try my cell phone from home tonight, but I suspect the problem has been solved. Just out of curiosity, is the relaxed mode looser than the DTMF spec or is it just the outside limits of the spec, and the normal mode is more restrictive than the DTMF spec? jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere DTMF solution?
Perhaps a clue ... with an established call, like: snom200<>*<>t400(daggressive_ecan_enabled)<>zhone<>speakerphone when the snom numberpad key is pressed the dtmf tone is not continuously robust but the volume is attenuated after the first ~1/10 sec for a moment and sort of resembles a double press. I have heard this with the different dtmfmodes. John - Original Message - From: "Mark Spencer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 12, 2003 6:51 PM Subject: Re: [Asterisk-Users] iconnecthere DTMF solution? > Probably you should do dtmfmode=inband in the general section. > > Mark > > On 12 Mar 2003, Matthew Farley wrote: > > > Finally, I have NATted ATA-186s working with Asterisk (thanks to > > all who made this happen)! My final troubles were with the firmware > > version in the 186 -- if you have troubles with this (as I did), make > > sure you have the newest firmware in the 186.. Otherwise it just won't > > work. > > > > Now for my current question - has anyone successfully gotten DTMF > > recognition to happen on iconnect SIP calls? I cannot seem to get it to > > work for either incoming or outgoing (tried the dtmfmode=inband in > > sip.conf, and am running the newest CVS version of Asterisk). I found a > > few messages on this topic in the archives, but did not find anything > > with a definitive answer. My apologies if this is question is redundant. > > If anyone knows how to get this working, I would be very grateful > > indeed. > > > > Sincerely, > > -- > > Matthew Farley <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music - Hold -> live sound source ?
On Wed, 12 Mar 2003 19:25:16 -0600, Steven Critchfield wrote: >On Wed, 2003-03-12 at 18:35, Gary wrote: >> Hi folks, >> >> I am looking for a way to actually have a live feed for a music on hold >> channel. >> >> Basically this is for a local radio station feed so people can ring to >> get the lastest report... Sort of like the US vhf weather channels >> >> ANy ideas please ?? > >If you are only wanting to use 1 line, then you could use chan_oss and >autoanswer to connect an incoming call to the soundcard. If you want >more, you may need to write an app that would access the soundcard or an >app that could multiplex the soundcards input. >-- >Steven Critchfield <[EMAIL PROTECTED]> it does need to service multiple lines at the same time. Also as a background music (on loudspeaker type phones) would be nice. We are currently playing with festival, so the caller can select which area they want details for and we hourly download and massage the hourly updates avail from our weather mob... . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music - Hold -> live sound source ?
On Wed, 2003-03-12 at 18:35, Gary wrote: > Hi folks, > > I am looking for a way to actually have a live feed for a music on hold > channel. > > Basically this is for a local radio station feed so people can ring to > get the lastest report... Sort of like the US vhf weather channels > > ANy ideas please ?? If you are only wanting to use 1 line, then you could use chan_oss and autoanswer to connect an incoming call to the soundcard. If you want more, you may need to write an app that would access the soundcard or an app that could multiplex the soundcards input. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960
Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music - Hold -> live sound source ?
Hi folks, I am looking for a way to actually have a live feed for a music on hold channel. Basically this is for a local radio station feed so people can ring to get the lastest report... Sort of like the US vhf weather channels ANy ideas please ?? Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Different ring pattern for extensions possible?
look at the 'r' option for the Dial command. Example exten => 333,1,Dial(Zap/1r2) --Mike On Wednesday, March 12, 2003, at 04:03 PM, Jim Archer wrote: Hi All... Is it possible to change the ringing pattern of an extension? I would like to make my extension ring differently based upon what number (distinctive ring or different channel) I am being called from. The best use of this I think would be to let internal calls ring differently than external calls. Also, I would like to give my family a number that only they know and that rings my extension in a special way. I would appreciate any suggestions! Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
version is 'Asterisk CVS-03/11/03-09:57:33' we can regist to wcom in two ways. first. register => masakazu:[EMAIL PROTECTED] * send REGISTER, but no response from wcom. second. quit * and change the way with number. like this. register => 9706052:[EMAIL PROTECTED] and REGISTER again. in this time,get this result following. mack*CLI> Interface is eth0 IP Address is 210.194.204.16 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:166.60.255.41 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f >From: ;tag=2d864abf To: ;tag=2d864abf Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 166.60.255.41:5060 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:0.0.0.0 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf >From: ;tag=2f12f9af To: ;tag=2f12f9af Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 0.0.0.0:0 XXX Need to handle Retransmitting XXX: REGISTER sip:166.60.255.41 SIP/2.0 Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f >From: ;tag=2d864abf To: ;tag=2d864abf Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration (no NAT) to 166.60.255.41:5060 and keep this trying. I sometime meets segfault.maybe that cause... --- Masakazu Nakano On Tue, 11 Mar 2003 22:34:30 -0600 (CST) Mark Spencer <[EMAIL PROTECTED]> wrote: >> **ASTERISK SIP PACKET >> >> XXX Need to handle Retransmitting XXX: >> REGISTER sip:166.60.255.41 SIP/2.0 >> Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924 >> From: ;tag=08e71f4b >> To: ;tag=08e71f4b >> Contact: >> Call-ID: [EMAIL PROTECTED] >> CSeq: 113 REGISTER >> User-Agent: Asterisk PBX >> Expires: 120 >> Event: registration >> (no NAT) to 166.60.255.41:5060 >> >> > >Do we not receive anything back at all? > >Mark > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold? Can't get it to work.
I just posted something earlier regarding music on hold not working.
I've got to believe that it is something I have done or am not doing but
have yet to figure out what.
Anyway have you checked your message and debug logs?
--
Walt Davis
www.waltdavis.net
> This is two SIP/7960 phones... when one is put on hold, the
> music on hold doesn't come to the other. I have the official
> mpg123 code installed (not the mpg321...). I do have the zaptel
> driver installed since I have a Wildcard FXO card in there
> for PSTN access...
>
> zapata.conf:
> musiconhold=random
>
> musiconhold.conf:
> random => quietmp3:/var/lib/asterisk/mohmp3,-z
>
> /var/lib/asterisk/mohmp3 does have the one sample mp3 file
> in there.
>
> And this works in extensions.conf:
>
> extensions.conf:
>
> exten => ,1,Answer ; Answer the line
> exten => ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
>
> Thoughts?
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
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[Asterisk-Users] Different ring pattern for extensions possible?
Hi All... Is it possible to change the ringing pattern of an extension? I would like to make my extension ring differently based upon what number (distinctive ring or different channel) I am being called from. The best use of this I think would be to let internal calls ring differently than external calls. Also, I would like to give my family a number that only they know and that rings my extension in a special way. I would appreciate any suggestions! Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring detection example?
Hi All... I read the discussion from last December about the various options for detecting distinctive ringing, but I could not find what the final decision was. Could someone please point me at an example of how to configure Asterisk to recognize distinctive ring? Thanks! Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 calling VM no DTMF ... Current CVScode broke.
> with latest CVS rfc2833 DTMF are not detected even when explicitly stated in > sip.conf. Need to do some debugging. This could be yet another side effect of the merger of Ross's code. From now on big contributed patches will need to have OK's from at least 3 people on the list who are testing the affected areas. Sound like a plan? > Plus, 'sip show channel ' segfaults asterisk. Fixed the segfault (doh!) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere DTMF solution?
Probably you should do dtmfmode=inband in the general section. Mark On 12 Mar 2003, Matthew Farley wrote: > Finally, I have NATted ATA-186s working with Asterisk (thanks to > all who made this happen)! My final troubles were with the firmware > version in the 186 -- if you have troubles with this (as I did), make > sure you have the newest firmware in the 186.. Otherwise it just won't > work. > > Now for my current question - has anyone successfully gotten DTMF > recognition to happen on iconnect SIP calls? I cannot seem to get it to > work for either incoming or outgoing (tried the dtmfmode=inband in > sip.conf, and am running the newest CVS version of Asterisk). I found a > few messages on this topic in the archives, but did not find anything > with a definitive answer. My apologies if this is question is redundant. > If anyone knows how to get this working, I would be very grateful > indeed. > > Sincerely, > -- > Matthew Farley <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interest in E1 channel banks?
Brendan, Can we catch up tommorrow before 3pm or around middle of the day on Tuesday next? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 calling VM no DTMF ... Current CVS code broke.
On Wednesday 12 March 2003 21:25, Jim Gottlieb wrote: > On 2003-03-12 at 12:43, you wrote: > > I had updated CVS this morning and it broke me being able > > to call the voicemail extension from my SIP/Cisco 7960 phone > > it won't receive DTMF digits... > > I noticed this last night and found I could fix it by adding > > dtmfmode=rfc2833 > > into each extension definition in sip.conf. with latest CVS rfc2833 DTMF are not detected even when explicitly stated in sip.conf. Plus, 'sip show channel ' segfaults asterisk. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
Are you running latest CVS? I think we addressed this issue a few days ago with the hit='f' thing. Mark On Wed, 12 Mar 2003, Darrell Eldridge wrote: > I doubt it's a signal loss problem. It's a simple > circuit, connected through our local Meridian, so it's > fax-to-Meridian [A] > Meridian internal [D] > Meridian to Channel Bank [A] > Channel Bank to Asterisk [D] > And I know that the fax works okay; it's a production > unit that gets used every day. > > As for the conf's, I've tried several things. Here's > my best guess so far: > > from zapata.conf: > > context => incomingfax > channel => 47 > > and from extensions.conf: > > [incomingfax] > exten => s,1,Answer > exten => s,2,Wait,18 > exten => s,3,Goto,incomingmain|s|1 ; roll to main > exten => fax,Dial,Zap/3 > > Fax calls are answered, then * waits 18 seconds (while > the fax machine is beeping) then * rolls the call to > incomingmain|s|1. > > Ideas? > > --- Jon Pounder <[EMAIL PROTECTED]> wrote: > > > > I know there was discussion at one point of signal > > loss through multiple > > d/a and a/d conversions > > what speed does it connect at through the extension > > ? (assuming analog line in) > > > > Tim - can you show us your config as an example ? > > > > > > At 11:29 AM 3/12/2003 -0800, you wrote: > > > > > > > >I still haven't been able to get fax detection > > going, > > > >but I came across something: when I execute "zap > > show > > > >channel 47" one of the parameters shown is "Fax > > > >Handled: no". I assume that's a reflection of > > > >something in zapata.conf, but I don't find > > anything > > > >there. Should it read "...yes" in order for > > Asterisk > > > >to detect the fax tones? If so, what's the > > syntax for > > > >setting it to yes? > > >I know this is working for me as of cvs march 9, 03 > > >as long as the fax machine calling you is sending > > a REAL CNG tone at 1100 > > >Hz > > > > > >can you show us the relevant context section in the > > extension.conf > > >ie the exten -> s entries & your fax extension > > entry > > > > > > > > >___ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > __ > Do you Yahoo!? > Yahoo! Web Hosting - establish your business online > http://webhosting.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
>And I know that the fax works okay; it's a production >unit that gets used every day. > >As for the conf's, I've tried several things. Here's >my best guess so far: > >from zapata.conf: > > context => incomingfax > channel => 47 > >and from extensions.conf: > > [incomingfax] > exten => s,1,Answer > exten => s,2,Wait,18 > exten => s,3,Goto,incomingmain|s|1 ; roll to main > exten => fax,Dial,Zap/3 > >Fax calls are answered, then * waits 18 seconds (while >the fax machine is beeping) then * rolls the call to >incomingmain|s|1. > >Ideas? try removing the wait & use a non-blocking background next I wonder why you need fax detect logic at all looks like you have a dedicated fax machine on a dedicated channel why don't you just send all [incomingFax] straight to Zap/3 [incomingfax] exten => s,1,Answer exten => s,2,Dial,Zap/3 and in zapata.conf context => incomingfax immediate=yes ;ans w/o waiting 2 rings then use ans in ext context Need usecallerid=no usecallerid=no channel => 47 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnecthere DTMF solution?
Finally, I have NATted ATA-186s working with Asterisk (thanks to all who made this happen)! My final troubles were with the firmware version in the 186 -- if you have troubles with this (as I did), make sure you have the newest firmware in the 186.. Otherwise it just won't work. Now for my current question - has anyone successfully gotten DTMF recognition to happen on iconnect SIP calls? I cannot seem to get it to work for either incoming or outgoing (tried the dtmfmode=inband in sip.conf, and am running the newest CVS version of Asterisk). I found a few messages on this topic in the archives, but did not find anything with a definitive answer. My apologies if this is question is redundant. If anyone knows how to get this working, I would be very grateful indeed. Sincerely, -- Matthew Farley <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
I doubt it's a signal loss problem. It's a simple circuit, connected through our local Meridian, so it's fax-to-Meridian [A] Meridian internal [D] Meridian to Channel Bank [A] Channel Bank to Asterisk [D] And I know that the fax works okay; it's a production unit that gets used every day. As for the conf's, I've tried several things. Here's my best guess so far: from zapata.conf: context => incomingfax channel => 47 and from extensions.conf: [incomingfax] exten => s,1,Answer exten => s,2,Wait,18 exten => s,3,Goto,incomingmain|s|1 ; roll to main exten => fax,Dial,Zap/3 Fax calls are answered, then * waits 18 seconds (while the fax machine is beeping) then * rolls the call to incomingmain|s|1. Ideas? --- Jon Pounder <[EMAIL PROTECTED]> wrote: > > I know there was discussion at one point of signal > loss through multiple > d/a and a/d conversions > what speed does it connect at through the extension > ? (assuming analog line in) > > Tim - can you show us your config as an example ? > > > At 11:29 AM 3/12/2003 -0800, you wrote: > > > > >I still haven't been able to get fax detection > going, > > >but I came across something: when I execute "zap > show > > >channel 47" one of the parameters shown is "Fax > > >Handled: no". I assume that's a reflection of > > >something in zapata.conf, but I don't find > anything > > >there. Should it read "...yes" in order for > Asterisk > > >to detect the fax tones? If so, what's the > syntax for > > >setting it to yes? > >I know this is working for me as of cvs march 9, 03 > >as long as the fax machine calling you is sending > a REAL CNG tone at 1100 > >Hz > > > >can you show us the relevant context section in the > extension.conf > >ie the exten -> s entries & your fax extension > entry > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Web Hosting - establish your business online http://webhosting.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
>Tim - can you show us your config as an example ? [inBound] exten => s,1,Answer exten => s,2,setmusiconhold,default exten => s,3,DigitTimeout,5 exten => s,4,responsetimeout,20 exten => s,5,BackGround,officemenu ;/var/lib/asterisk/sounds ;... other exts ;Fax Test exten => fax,1,Dial,Zap/6 -- If you are not getting a fax re-route when using this set up then you prolly need to check & see if the logic in dsp.d->dtmf_detect lines 510-540 are seeing a CNG tone based on the fax_energy var look for this test if (!hit && (fax_energy >= FAX_THRESHOLD) && (fax_energy > s->energy * 21.0)) and add some debugging to see what fax_energy & s-energy look like sumfin like ast_log(LOG_DEBUG, "fax_energy %f ratio %f \n", fax_energy, fax_energy/s-energy ); Also when playing with this last week & googling around it looks like the ITU allows 4 standards for a fax handshake, only one of which is for the sending fax device to start the hand shake with a CNG tone. I beleive to have discovered that my telephone/fax for example is one of these devices that will call a fax device then wait for the called fax device to send a CED tone (2100hz) then start the fax transmisison only after the called device sends the CED.. in this case * will fail to detect the remote fax bcus * does not send the CED tone, during the ans sequence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain settings
I cranked them up around 15 and now the voice levels appear to match the levels for the automatic voice (like the voice mail and the directory). Doing that seems to distort the caller id so Asterisk can't decode it. I'm still experimenting. Thanks! --On Wednesday, March 12, 2003 8:55 PM + T Aksoy <[EMAIL PROTECTED]> wrote: rxgain and txgain are in db. We have a similar problem which is even more noticeable since we divert calls by receiving on one fxo card #1 and sending out on fxo card #2. I can't seem to find a properly working solution for the attentuation which is taking place. For your issue, try setting txgain to around 6.0 and see if it's any better. I think that you will need to restart asterisk for settings to take effect. Tan - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 12, 2003 2:06 AM Subject: [Asterisk-Users] Gain settings Hi All... I am using Asterisk on Debian with a single FXO card. I find that when I dial into it it sounds very soft. I also noticed that when I record VM greetings (I use the USB device for FXS) they are very soft. I saw the rxgain and txgain. Can some one tell me how these are used? I have seen examples with 0.0 and others with 100%. I have played with integers and found they can make it very loud and distorted. Whats the proper system? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain settings
rxgain and txgain are in db. We have a similar problem which is even more noticeable since we divert calls by receiving on one fxo card #1 and sending out on fxo card #2. I can't seem to find a properly working solution for the attentuation which is taking place. For your issue, try setting txgain to around 6.0 and see if it's any better. I think that you will need to restart asterisk for settings to take effect. Tan - Original Message - From: "Jim Archer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 12, 2003 2:06 AM Subject: [Asterisk-Users] Gain settings Hi All... I am using Asterisk on Debian with a single FXO card. I find that when I dial into it it sounds very soft. I also noticed that when I record VM greetings (I use the USB device for FXS) they are very soft. I saw the rxgain and txgain. Can some one tell me how these are used? I have seen examples with 0.0 and others with 100%. I have played with integers and found they can make it very loud and distorted. Whats the proper system? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
On Wed, 2003-03-12 at 14:32, Martin Pycko wrote: > You may try to add > relaxdtmf=yes > just before channel => 4 in zapata.conf Thanks! This has solved the problem for the test phone! I will try my cell phone from home tonight, but I suspect the problem has been solved. Just out of curiosity, is the relaxed mode looser than the DTMF spec or is it just the outside limits of the spec, and the normal mode is more restrictive than the DTMF spec? jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 calling VM no DTMF ... Current CVS code broke.
On 2003-03-12 at 12:43, you wrote: > I had updated CVS this morning and it broke me being able > to call the voicemail extension from my SIP/Cisco 7960 phone > it won't receive DTMF digits... I noticed this last night and found I could fix it by adding dtmfmode=rfc2833 into each extension definition in sip.conf. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
I tried that, same thing. Cell phones horrible, but landlines work fine. On Wednesday, March 12, 2003, at 02:59 PM, Mark Spencer wrote: there is a relaxed dtmf mode that may help. Mark On 12 Mar 2003, James Hines wrote: On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote: I am using background, the pbx-invalid stuff should (if DTMF recognition is working correctly) not get played. My users here have complained about similar problems. We've noticed it most often on outside callers with cell phones, but one user brought in a phone from home that pretty consistently would mess up. I hooked it up to an inside line and it still happens. My guess is that asterisk is being a little to stringent with it's DTMF detection. Looking at the log asterisk will detect multiple DTMF events for a single key press. I looked at a recording of the tone and it looked a bit different than a phone that generated 'good' tones 100% of the time, but it certainly wasn't sending a tone and stopping. jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
Exactly! Every test I have done has been with a cell phone! I assume everyone is still perplexed by this? On Wednesday, March 12, 2003, at 02:12 PM, James Hines wrote: On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote: I am using background, the pbx-invalid stuff should (if DTMF recognition is working correctly) not get played. My users here have complained about similar problems. We've noticed it most often on outside callers with cell phones, but one user brought in a phone from home that pretty consistently would mess up. I hooked it up to an inside line and it still happens. My guess is that asterisk is being a little to stringent with it's DTMF detection. Looking at the log asterisk will detect multiple DTMF events for a single key press. I looked at a recording of the tone and it looked a bit different than a phone that generated 'good' tones 100% of the time, but it certainly wasn't sending a tone and stopping. jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF Digits
I have a fork for this thread. With our asterisk system we have to enter my touchpad keys a little more carefully than on other systems. Is there parameters to adjust this behavior. Brian, this is probably your problem... >> exten => s,1,Answer >> exten => s,2,Wait,1 >> exten => s,3,DigitTimeout,10 Move up or eliminate wait ;>> exten => s,2,Wait,1 >> exten => s,1,Answer >> exten => s,3,DigitTimeout,10 John Harragin This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
there is a relaxed dtmf mode that may help. Mark On 12 Mar 2003, James Hines wrote: > On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote: > > I am using background, the pbx-invalid stuff should (if DTMF > > recognition is working correctly) not get played. > > > > My users here have complained about similar problems. We've noticed it > most often on outside callers with cell phones, but one user brought in > a phone from home that pretty consistently would mess up. I hooked it up > to an inside line and it still happens. My guess is that asterisk is > being a little to stringent with it's DTMF detection. Looking at the log > asterisk will detect multiple DTMF events for a single key press. I > looked at a recording of the tone and it looked a bit different than a > phone that generated 'good' tones 100% of the time, but it certainly > wasn't sending a tone and stopping. > > jwsh > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 calling VM no DTMF ... Current CVScode broke.
how does your cisco send DTMF? Mark On Wed, 12 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote: > I had updated CVS this morning and it broke me being able > to call the voicemail extension from my SIP/Cisco 7960 phone > it won't receive DTMF digits... > > Restored back to Mar 10 2003 and it worked just fine... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working
Same for me , when I call from one sip fxs gw phone to the snom one. I can hear audio only on from the sip gw and not from the snom. Thery're using only alaw/ulaw (only accepted in sip.conf). Yesterday all was working. >From sip to zap or viceversa is all ok. Need a debug? Matteo > -Messaggio originale- > Da: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Per conto di > Lubomir Christov > Inviato: mercoledì 12 marzo 2003 19.01 > A: [EMAIL PROTECTED]; [EMAIL PROTECTED] > Oggetto: [Asterisk-Users] SIP/G723/iconnect with todays CVS > version isn't working > > > Hello all, > > I'm using iconnect with LineJACK/PhoneJACK/PhoneCARD and G723.1 codec > from about 1 mount without any problems. The quality is ok and > everything is OK (only some little problems sometime ... when > the format > in phone.conf isn't slinear, but format=g723.1 I have only > ONE way audio > (the other side is hearing ONLY strange sounds )). > But today morning, when I updated new CVS version of * I found that > SIP(G723/ulaw) and iconnect aren't working anymore ??? > > When I try to connect trough iconnect I receive this error message: > > -- Got SIP response 488 "Not Acceptable Media" back from > 213.137.73.178 > > here is my config: > > sip.conf > [general] > port = 5060 > ;bindaddr = 0.0.0.0 > context = incoming > disallow=all > allow=g723.1 > ;allow=ulaw > tos=lowdelay > tos=184 > > [iconnect] > type=friend > username=12345678 > password=1234 > host=213.137.73.178 > callerid=1234567890 > > I have attached my todays sip debug output. > I'm sure that the problem is in todays CVS version only > because when I > download yesterdays version (cvs -z9 co -D "Mar 11 2003" > asterisk) there > wasn't such a problem and everything was OK. > I hope that iconnect will be back soon :))) > > Lubo > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
You may try to add relaxdtmf=yes just before channel => 4 in zapata.conf regards Martin On Wed, 12 Mar 2003, Brian J. Schrock wrote: > I am using background, the pbx-invalid stuff should (if DTMF > recognition is working correctly) not get played. > > On Wednesday, March 12, 2003, at 01:30 PM, Steven Critchfield wrote: > > > Playback is not interuptable, use Background. > > > > On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote: > >> Hello, > >> > >> I am having a problem with Asterisk that I just cannot get fixed... > >> When I call in to the main number I have to wait until well into the > >> second message shown in the extensions.conf snippet below to enter an > >> extension number. If I enter digits really slowly sometimes it will > >> work during the first message. Usually my callers just get the > >> incredibly annoying "invalid message recording". I remember reading on > >> here about DTMF detection problems earlier, but cannot find anything > >> relevant. Has anyone else had this problem, or does anyone else know > >> what could be the problem? > >> > >> My extension are all 2244,2245,and 2246. If I enter 2244 too early in > >> the playback Asterisk recognizes it as 222 or sometimes 24. > >> > >> If I unload the zap drivers and reload them and restart asterisk it > >> will work just fine for the first call but after that the problem > >> shows > >> up. > >> > >> If I play with txgain and rxgain enough I can make the problem worse > >> but not better. > >> > >> My asterisk is from cvs two days ago. > >> > >> I have been turning echo cancellation on and off in different > >> combinations to see how it affects everything, and it did not have an > >> impact. > >> > >> ##Extensions.conf## > >> > >> [Afternoon] > >> exten => t,1,Goto,default|s|1 > >> exten => i,1,Playback,pbx-invalid > >> exten => i,2,Goto,default|s|1 > >> include => extensions > >> exten => s/_6145551234,1,Answer > >> exten => s/_6145551234,2,Dial,Zap/g2 > >> exten => s,1,Answer > >> exten => s,2,Wait,1 > >> exten => s,3,DigitTimeout,10 > >> exten => s,4,Background,Afternoon_Intro > >> exten => s,5,Background,Exten_Direct > >> > >> ##Zapata.conf## > >> > >> [channels] > >> > >> txgain=0 > >> rxgain=0 > >> context = default > >> language = en > >> callwaiting = yes > >> callwaitingcallerid = no > >> threewaycalling = yes > >> transfer = yes > >> cancelforward = yes > >> callreturn = no > >> usecallerid = yes > >> hidecallerid = no > >> echocancel = yes > >> echocancelwhenbridged = no > >> immediate = yes > >> > >> group = 1 > >> ;use with FXO PCI card > >> signalling = fxs_ks > >> channel => 1-3 > >> > >> echocancel = yes > >> echocancelwhenbridged = yes > >> context = local > >> immediate = no > >> group = 2 > >> txgain=0 > >> rxgain=0 > >> ;use with FXS USB card > >> signalling = fxo_ks > >> callerid = "Brian Schrock" <(614) 798-9106> > >> mailbox=2244,2245,2246 > >> channel => 4 > >> > >> Brian J. Schrock > >> Network Engineer, RHCE, CCNA > >> Anistone Technologies > >> Phone: 614-798-9106 > >> FAX: 614-573-7165 > >> 6926 Avery Rd. > >> Dublin, OH 43017 > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-798-9106 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
I know there was discussion at one point of signal loss through multiple d/a and a/d conversions what speed does it connect at through the extension ? (assuming analog line in) Tim - can you show us your config as an example ? At 11:29 AM 3/12/2003 -0800, you wrote: >I still haven't been able to get fax detection going, >but I came across something: when I execute "zap show >channel 47" one of the parameters shown is "Fax >Handled: no". I assume that's a reflection of >something in zapata.conf, but I don't find anything >there. Should it read "...yes" in order for Asterisk >to detect the fax tones? If so, what's the syntax for >setting it to yes? I know this is working for me as of cvs march 9, 03 as long as the fax machine calling you is sending a REAL CNG tone at 1100 Hz can you show us the relevant context section in the extension.conf ie the exten -> s entries & your fax extension entry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
It's dynamically changed to "Yes" when the fax gets detected on this channel. regards Martin On Wed, 12 Mar 2003, Darrell Eldridge wrote: > I still haven't been able to get fax detection going, > but I came across something: when I execute "zap show > channel 47" one of the parameters shown is "Fax > Handled: no". I assume that's a reflection of > something in zapata.conf, but I don't find anything > there. Should it read "...yes" in order for Asterisk > to detect the fax tones? If so, what's the syntax for > setting it to yes? > > __ > Do you Yahoo!? > Yahoo! Web Hosting - establish your business online > http://webhosting.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
>I still haven't been able to get fax detection going, >but I came across something: when I execute "zap show >channel 47" one of the parameters shown is "Fax >Handled: no". I assume that's a reflection of >something in zapata.conf, but I don't find anything >there. Should it read "...yes" in order for Asterisk >to detect the fax tones? If so, what's the syntax for >setting it to yes? I know this is working for me as of cvs march 9, 03 as long as the fax machine calling you is sending a REAL CNG tone at 1100 Hz can you show us the relevant context section in the extension.conf ie the exten -> s entries & your fax extension entry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote: > I am using background, the pbx-invalid stuff should (if DTMF > recognition is working correctly) not get played. > My users here have complained about similar problems. We've noticed it most often on outside callers with cell phones, but one user brought in a phone from home that pretty consistently would mess up. I hooked it up to an inside line and it still happens. My guess is that asterisk is being a little to stringent with it's DTMF detection. Looking at the log asterisk will detect multiple DTMF events for a single key press. I looked at a recording of the tone and it looked a bit different than a phone that generated 'good' tones 100% of the time, but it certainly wasn't sending a tone and stopping. jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Fax Handled: no" config
Fax handled will be set to yes when zaptel detects the fax CNG tones on the line. At that point it will try to switch the call to the fax extension James On Wed, 12 Mar 2003, Darrell Eldridge wrote: > I still haven't been able to get fax detection going, > but I came across something: when I execute "zap show > channel 47" one of the parameters shown is "Fax > Handled: no". I assume that's a reflection of > something in zapata.conf, but I don't find anything > there. Should it read "...yes" in order for Asterisk > to detect the fax tones? If so, what's the syntax for > setting it to yes? > > __ > Do you Yahoo!? > Yahoo! Web Hosting - establish your business online > http://webhosting.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: [Asterisk-Users] Several patches, including recording and music-on-hold
A fake web Cvs repository is already on, checkout http://vmail.espia.it/horde/chora Unfortunately, due to the limitations of cvs, there's only 1 revision, that's 1.1 , and only 1 author, that's the user I upload as (matteo). I update it on a daily basis (07:00 am Rome Time). I use it to checkout if there're differences day by day. (pretty useful in that). To have a *real* web cvs viewer, only digium could fire up that service (since the web need access to the cvs dirs). Another solution is to have a rsync access to the cvs tree, in order to duplicate the cvs server dirs. I'll be glad to put up a real cvs viewer, if digium could give rsync readonly access to the cvs server dirs. Matteo. > -Messaggio originale- > Da: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Per conto di > Florian Overkamp > Inviato: mercoledì 12 marzo 2003 17.21 > A: [EMAIL PROTECTED] > Oggetto: Re: R: [Asterisk-Users] Several patches, including > recording and music-on-hold > > > Hi, > > At 10:30 12-3-2003 +0100, you wrote: > >(i'm thinking to put up a little website, > >with a repository of all * patches. tell me > >if anyone is interested into that) > > Yes, this is -very- usefull. > > How about a cvs repository :-) > > Best regards, > Florian > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold? Can't get it to work.
This is two SIP/7960 phones... when one is put on hold, the
music on hold doesn't come to the other. I have the official
mpg123 code installed (not the mpg321...). I do have the zaptel
driver installed since I have a Wildcard FXO card in there
for PSTN access...
zapata.conf:
musiconhold=random
musiconhold.conf:
random => quietmp3:/var/lib/asterisk/mohmp3,-z
/var/lib/asterisk/mohmp3 does have the one sample mp3 file
in there.
And this works in extensions.conf:
extensions.conf:
exten => ,1,Answer ; Answer the line
exten => ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
Thoughts?
___
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[Asterisk-Users] Cisco 7960 calling VM no DTMF ... Current CVS code broke.
I had updated CVS this morning and it broke me being able to call the voicemail extension from my SIP/Cisco 7960 phone it won't receive DTMF digits... Restored back to Mar 10 2003 and it worked just fine... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
I am using background, the pbx-invalid stuff should (if DTMF recognition is working correctly) not get played. On Wednesday, March 12, 2003, at 01:30 PM, Steven Critchfield wrote: Playback is not interuptable, use Background. On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote: Hello, I am having a problem with Asterisk that I just cannot get fixed... When I call in to the main number I have to wait until well into the second message shown in the extensions.conf snippet below to enter an extension number. If I enter digits really slowly sometimes it will work during the first message. Usually my callers just get the incredibly annoying "invalid message recording". I remember reading on here about DTMF detection problems earlier, but cannot find anything relevant. Has anyone else had this problem, or does anyone else know what could be the problem? My extension are all 2244,2245,and 2246. If I enter 2244 too early in the playback Asterisk recognizes it as 222 or sometimes 24. If I unload the zap drivers and reload them and restart asterisk it will work just fine for the first call but after that the problem shows up. If I play with txgain and rxgain enough I can make the problem worse but not better. My asterisk is from cvs two days ago. I have been turning echo cancellation on and off in different combinations to see how it affects everything, and it did not have an impact. ##Extensions.conf## [Afternoon] exten => t,1,Goto,default|s|1 exten => i,1,Playback,pbx-invalid exten => i,2,Goto,default|s|1 include => extensions exten => s/_6145551234,1,Answer exten => s/_6145551234,2,Dial,Zap/g2 exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,DigitTimeout,10 exten => s,4,Background,Afternoon_Intro exten => s,5,Background,Exten_Direct ##Zapata.conf## [channels] txgain=0 rxgain=0 context = default language = en callwaiting = yes callwaitingcallerid = no threewaycalling = yes transfer = yes cancelforward = yes callreturn = no usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = no immediate = yes group = 1 ;use with FXO PCI card signalling = fxs_ks channel => 1-3 echocancel = yes echocancelwhenbridged = yes context = local immediate = no group = 2 txgain=0 rxgain=0 ;use with FXS USB card signalling = fxo_ks callerid = "Brian Schrock" <(614) 798-9106> mailbox=2244,2245,2246 channel => 4 Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960/SIP put on hold when returned can't hear...
A bug that I've been meaning to report... When you call someone and have the remote person put you on hold (both are Cisco 7960/SIP recipients), when they come back off of hold they can hear me, but I cannot hear them... sounds like one of the audio channels is not restored properly.. I'll be happy to debug if someone needs more debug output. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Several patches, including recording and music-on-hold
Florian Overkamp wrote: > Hi, > > At 10:30 12-3-2003 +0100, you wrote: >>(i'm thinking to put up a little website, >>with a repository of all * patches. tell me >>if anyone is interested into that) > > Yes, this is -very- usefull. > > How about a cvs repository :-) it would also be nice that a site have patch, music, config file example and documentation. Thanks, Jean-Pierre Denis jp at msfree dot ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "Fax Handled: no" config
I still haven't been able to get fax detection going, but I came across something: when I execute "zap show channel 47" one of the parameters shown is "Fax Handled: no". I assume that's a reflection of something in zapata.conf, but I don't find anything there. Should it read "...yes" in order for Asterisk to detect the fax tones? If so, what's the syntax for setting it to yes? __ Do you Yahoo!? Yahoo! Web Hosting - establish your business online http://webhosting.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Digits
Playback is not interuptable, use Background. On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote: > Hello, > > I am having a problem with Asterisk that I just cannot get fixed... > When I call in to the main number I have to wait until well into the > second message shown in the extensions.conf snippet below to enter an > extension number. If I enter digits really slowly sometimes it will > work during the first message. Usually my callers just get the > incredibly annoying "invalid message recording". I remember reading on > here about DTMF detection problems earlier, but cannot find anything > relevant. Has anyone else had this problem, or does anyone else know > what could be the problem? > > My extension are all 2244,2245,and 2246. If I enter 2244 too early in > the playback Asterisk recognizes it as 222 or sometimes 24. > > If I unload the zap drivers and reload them and restart asterisk it > will work just fine for the first call but after that the problem shows > up. > > If I play with txgain and rxgain enough I can make the problem worse > but not better. > > My asterisk is from cvs two days ago. > > I have been turning echo cancellation on and off in different > combinations to see how it affects everything, and it did not have an > impact. > > ##Extensions.conf## > > [Afternoon] > exten => t,1,Goto,default|s|1 > exten => i,1,Playback,pbx-invalid > exten => i,2,Goto,default|s|1 > include => extensions > exten => s/_6145551234,1,Answer > exten => s/_6145551234,2,Dial,Zap/g2 > exten => s,1,Answer > exten => s,2,Wait,1 > exten => s,3,DigitTimeout,10 > exten => s,4,Background,Afternoon_Intro > exten => s,5,Background,Exten_Direct > > ##Zapata.conf## > > [channels] > > txgain=0 > rxgain=0 > context = default > language = en > callwaiting = yes > callwaitingcallerid = no > threewaycalling = yes > transfer = yes > cancelforward = yes > callreturn = no > usecallerid = yes > hidecallerid = no > echocancel = yes > echocancelwhenbridged = no > immediate = yes > > group = 1 > ;use with FXO PCI card > signalling = fxs_ks > channel => 1-3 > > echocancel = yes > echocancelwhenbridged = yes > context = local > immediate = no > group = 2 > txgain=0 > rxgain=0 > ;use with FXS USB card > signalling = fxo_ks > callerid = "Brian Schrock" <(614) 798-9106> > mailbox=2244,2245,2246 > channel => 4 > > Brian J. Schrock > Network Engineer, RHCE, CCNA > Anistone Technologies > Phone: 614-798-9106 > FAX: 614-573-7165 > 6926 Avery Rd. > Dublin, OH 43017 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Digits
Hello, I am having a problem with Asterisk that I just cannot get fixed... When I call in to the main number I have to wait until well into the second message shown in the extensions.conf snippet below to enter an extension number. If I enter digits really slowly sometimes it will work during the first message. Usually my callers just get the incredibly annoying "invalid message recording". I remember reading on here about DTMF detection problems earlier, but cannot find anything relevant. Has anyone else had this problem, or does anyone else know what could be the problem? My extension are all 2244,2245,and 2246. If I enter 2244 too early in the playback Asterisk recognizes it as 222 or sometimes 24. If I unload the zap drivers and reload them and restart asterisk it will work just fine for the first call but after that the problem shows up. If I play with txgain and rxgain enough I can make the problem worse but not better. My asterisk is from cvs two days ago. I have been turning echo cancellation on and off in different combinations to see how it affects everything, and it did not have an impact. ##Extensions.conf## [Afternoon] exten => t,1,Goto,default|s|1 exten => i,1,Playback,pbx-invalid exten => i,2,Goto,default|s|1 include => extensions exten => s/_6145551234,1,Answer exten => s/_6145551234,2,Dial,Zap/g2 exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,DigitTimeout,10 exten => s,4,Background,Afternoon_Intro exten => s,5,Background,Exten_Direct ##Zapata.conf## [channels] txgain=0 rxgain=0 context = default language = en callwaiting = yes callwaitingcallerid = no threewaycalling = yes transfer = yes cancelforward = yes callreturn = no usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = no immediate = yes group = 1 ;use with FXO PCI card signalling = fxs_ks channel => 1-3 echocancel = yes echocancelwhenbridged = yes context = local immediate = no group = 2 txgain=0 rxgain=0 ;use with FXS USB card signalling = fxo_ks callerid = "Brian Schrock" <(614) 798-9106> mailbox=2244,2245,2246 channel => 4 Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Please check latest CVS. This issue has been fixed and was related to the
dynamic payload merger.
Mark
On Wed, 12 Mar 2003, Lubomir Christov wrote:
> Hi Gregg,
>
> I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
> mount without any problems. The quality is perfect and everything is OK
> (only some little problems sometime).
> But today morning, with the NEW CVS version update of asterisk I found
> that SIP(G723/ulaw) and iconnect aren't working anymore ???
> When I try to connect trough iconnect I receive this error message:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178
>
> you can try asterisk from yesterday:
>cvs -z9 co -D "Mar 11 2003" asterisk
>
> and test it: everything will be OK :)
> Here is my working configuration:
>
> sip.conf
>
> [general]
> port = 5060
> ;bindaddr = 0.0.0.0
> context = incomming
> disallow=all
> allow=g723.1
> ;allow=ulaw
> tos=lowdelay
> tos=184
>
> [iconnect]
> type=friend
> username=12345678
> password=1234
> host=213.137.73.178
> callerid=1234567890
>
>
>
> phone.conf
>
> format=slinear
> echocancel=low
> silencesupression=no
>
>
> extension.conf
>
> exten => _00.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,C)
>
> Lubo
>
> P.S. for successfully using G723 codec and phonejack you will need
> g723.1 and g723.1b placed in your codecs directory. You can got it like
> this:
>
> cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1
> cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1b
>
> and uncomment this line in Makefile in codecs directory
> MODG723=codec_g723_1.so codec_g723_1b.so
>
> I hope that the todays problem with asterisk and SIP/G723 will be fixed
> very soon.
>
> L
>
> Gregg Lebovitz wrote:
> > Hi Lubo,
> >
> > I appreciate your email to help with this issue, but I don't understand
> > your message. I assume your comment about format=slinear is to use
> > format=slinear in phone.conf instead of format=ulaw. If so, how does
> > this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
> >
> > Gregg
> >
> > On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote:
> >
> >>Dan, why are you using phonejack with ulaw codec? g723 (format=slinear
> >>only) is working just perfect with phonejack and iconnect :)
> >>
> >>Lubo
> >>
> >>Dan Fernandez wrote:
> >>
> >>>I found similar problems.
> >>>
> >>>With my phonejack I can make a call with ulaw with decent quality (I have a
> >>>64k line).
> >>>
> >>>However, with Messenger I hear a brief horrible noise and thats it.
> >>>
> >>>- Original Message -
> >>>From: "Jim Archer" <[EMAIL PROTECTED]>
> >>>To: <[EMAIL PROTECTED]>
> >>>Sent: Tuesday, March 11, 2003 8:17 PM
> >>>Subject: Re: [Asterisk-Users] iconnect quality?
> >>>
> >>>
> >>>
> >>>
> Ok! When I use the prefix and I allow gsm it does work! And the
> quality is fine.
>
> There are two problems we're having now.
>
> 1 - From watching the udp fly by, it seems that iconnect does not know
> >>>
> >>>when
> >>>
> >>>
> we hang up. For example, if I call a voice mail and it starts giving me
> its speal and I hang up, iconnect stays connected until the VM hangs up at
> its end.
>
> Next, if we try to call out via iconnect from a sip client extension (like
> a windows soft phone) all we hear is horrible noise.
>
> Has anyone else had these issues?
>
> Jim
>
>
> --On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz
> <[EMAIL PROTECTED]> wrote:
>
>
>
> >I haven't play around enough to know whether or not the prefix is a
> >toggle. I will do some experimenting and let you know. Right now I am
> >prefixing all my calls with .
> >
> >My experience is that when the carrier's format is G723.1, you can't
> >hear the incoming voice. When it is in G711 you can. I have made several
> >calls using G711 and they are acceptable quality. Note that if you
> >disallow=gsm in the sip.conf file you will get the 488 media errors you
> >reported earlier.
> >
> >Below are my config files for sip and the linejack cards:
> >
> >;
> >; SIP Configuration for Asterisk
> >;
> >[general]
> >port = 5060 ; Port to bind to
> >bindaddr = 0.0.0.0 ; Address to bind to
> >context=iconnect ; Default for incoming calls
> >allow=gsm
> >allow=ulaw
> >allow=alaw
> >
> >;register=1813342:[EMAIL PROTECTED]
> >;register=1202454:[EMAIL PROTECTED]
> >
> >[iconnecthere]
> >type=friend
> >username=
> >secret=XXX
> >host=sipauth.deltathree.com
> >
> >;
> >; Linux Telephony Interface
> >;
> >; Configuration file
> >;
> >[interfaces]
> >
> >mode=dialtone
> >format=ulaw
> >echocancel=medium
> >silencesupression=no
> >
> >context=local
> >context=default
> >
> >txgain=100%
> >rxgain=100%
> >device =
Re: [Asterisk-Users] ATA beginners question
This issue should be fixed now.
Mark
On Wed, 12 Mar 2003, Michiel Betel wrote:
> When dialing a port on my ATA-186 I get:
>
> == Spawn extension (default, s, 1) exited non-zero on 'SIP/ata1-1-0c77'
> -- Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
> stack
> -- Executing Dial("SIP/ata1-2-4fc0", "SIP/ata1-1|30") in new stack
> -- Called ata1-1
> -- Got SIP response 488 "Not Acceptable Here" back from 192.168.1.100
> == No one is available to answer at this time
>
> Instead of a ringing telephone... Both ata1-1 and ata1-2 are registered
> with asterisk.
>
> Can anyone tell me how to get rid of the 488?
>
> Thanks!
>
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working
Hello all,
I'm using iconnect with LineJACK/PhoneJACK/PhoneCARD and G723.1 codec
from about 1 mount without any problems. The quality is ok and
everything is OK (only some little problems sometime ... when the format
in phone.conf isn't slinear, but format=g723.1 I have only ONE way audio
(the other side is hearing ONLY strange sounds )).
But today morning, when I updated new CVS version of * I found that
SIP(G723/ulaw) and iconnect aren't working anymore ???
When I try to connect trough iconnect I receive this error message:
-- Got SIP response 488 "Not Acceptable Media" back from
213.137.73.178
here is my config:
sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
context = incoming
disallow=all
allow=g723.1
;allow=ulaw
tos=lowdelay
tos=184
[iconnect]
type=friend
username=12345678
password=1234
host=213.137.73.178
callerid=1234567890
I have attached my todays sip debug output.
I'm sure that the problem is in todays CVS version only because when I
download yesterdays version (cvs -z9 co -D "Mar 11 2003" asterisk) there
wasn't such a problem and everything was OK.
I hope that iconnect will be back soon :)))
Lubo
*CLI>
Sip read:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=0efccd4b
From: "asterisk" ;tag=545f9b5c
To: sip:[EMAIL PROTECTED];tag=D7559CE0-3D4
Date: Wed, 12 Mar 2003 17:11:22 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0
9 headers, 0 lines
Interface is ppp0
IP Address is 213.16.62.32
*CLI>
*CLI>
-- Executing Dial("Phone/phone0", "Sip/[EMAIL PROTECTED]||C") in new stack
Interface is ppp0
IP Address is 213.16.62.32
We're at 213.16.62.32 port 7562
Answering with preferred capability 1
10 headers, 6 lines
XXX Need to handle Retransmitting XXX:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
Contact:
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 106
v=0
o=root 4008 4008 IN IP4 213.16.62.32
s=session
c=IN IP4 213.16.62.32
t=0 0
m=audio 7562 RTP/AVP
(no NAT) to 213.137.73.178:5060
-- Called [EMAIL PROTECTED]
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: [EMAIL PROTECTED]
From: "asterisk" ;tag=5d3317c9
To:
CSeq: 102 INVITE
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: [EMAIL PROTECTED]
From: "asterisk" ;tag=5d3317c9
To: ;tag=534dad39-5e7fe66
CSeq: 102 INVITE
Proxy-Authenticate: DIGEST realm="deltathree.com", nonce="3e6f6a54", algorithm=MD5
Content-Length: 0
8 headers, 0 lines
XXX Need to handle Retransmitting XXX:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
To: ;tag=534dad39-5e7fe66
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 213.137.73.178:5060
We're at 213.16.62.32 port 7562
Answering with preferred capability 1
XXX Need to handle Retransmitting XXX:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
Contact:
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="12345678", realm="deltathree.com",
algorithm="MD5", uri="sip:[EMAIL PROTECTED]", nonce="3e6f6a54",
response="8863cae6c56b333a2c09b76e2a3013b3"
Content-Type: application/sdp
Content-Length: 106
v=0
o=root 3530 3530 IN IP4 213.16.62.32
s=session
c=IN IP4 213.16.62.32
t=0 0
m=audio 7562 RTP/AVP
(no NAT) to 213.137.73.178:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: [EMAIL PROTECTED]
From: "asterisk" ;tag=5d3317c9
To:
CSeq: 103 INVITE
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
To: sip:[EMAIL PROTECTED];tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0
9 headers, 0 lines
-- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178
XXX Need to handle Retransmitting XXX:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
To: ;tag=534dad39-5e7fe66
Call-ID: [EMAIL PROTECTED]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 213.137.73.178:5060
WARNING[245774]: File app_dial.c, Line 271 (wait_for_answer): Unable to forward voice
Sip read:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" ;tag=5d3317c9
To: sip:[EMAIL PROTECTED];tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103
Re: [Asterisk-Users] ATA beginners question
On Wed, Mar 12, 2003 at 05:35:44PM +0100, Michiel Betel wrote:
* from today cvs ?
May be same problem about my posting.
Asterisk don't send 'a' parameters SDP body SIP message to called SIP device.
I got same error (488) when using today cvs version.
Try cvs update -D '1 day ago'
On my * box it work properly.
> When dialing a port on my ATA-186 I get:
>
> == Spawn extension (default, s, 1) exited non-zero on 'SIP/ata1-1-0c77'
> -- Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
> stack
> -- Executing Dial("SIP/ata1-2-4fc0", "SIP/ata1-1|30") in new stack
> -- Called ata1-1
> -- Got SIP response 488 "Not Acceptable Here" back from 192.168.1.100
> == No one is available to answer at this time
>
> Instead of a ringing telephone... Both ata1-1 and ata1-2 are registered
> with asterisk.
>
> Can anyone tell me how to get rid of the 488?
>
> Thanks!
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Hi Gregg,
I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
mount without any problems. The quality is perfect and everything is OK
(only some little problems sometime).
But today morning, with the NEW CVS version update of asterisk I found
that SIP(G723/ulaw) and iconnect aren't working anymore ???
When I try to connect trough iconnect I receive this error message:
-- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178
you can try asterisk from yesterday:
cvs -z9 co -D "Mar 11 2003" asterisk
and test it: everything will be OK :)
Here is my working configuration:
sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
context = incomming
disallow=all
allow=g723.1
;allow=ulaw
tos=lowdelay
tos=184
[iconnect]
type=friend
username=12345678
password=1234
host=213.137.73.178
callerid=1234567890
phone.conf
format=slinear
echocancel=low
silencesupression=no
extension.conf
exten => _00.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,C)
Lubo
P.S. for successfully using G723 codec and phonejack you will need
g723.1 and g723.1b placed in your codecs directory. You can got it like
this:
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1
cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co g723.1b
and uncomment this line in Makefile in codecs directory
MODG723=codec_g723_1.so codec_g723_1b.so
I hope that the todays problem with asterisk and SIP/G723 will be fixed
very soon.
L
Gregg Lebovitz wrote:
Hi Lubo,
I appreciate your email to help with this issue, but I don't understand
your message. I assume your comment about format=slinear is to use
format=slinear in phone.conf instead of format=ulaw. If so, how does
this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
Gregg
On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote:
Dan, why are you using phonejack with ulaw codec? g723 (format=slinear
only) is working just perfect with phonejack and iconnect :)
Lubo
Dan Fernandez wrote:
I found similar problems.
With my phonejack I can make a call with ulaw with decent quality (I have a
64k line).
However, with Messenger I hear a brief horrible noise and that?s it.
- Original Message -
From: "Jim Archer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 11, 2003 8:17 PM
Subject: Re: [Asterisk-Users] iconnect quality?
Ok! When I use the prefix and I allow gsm it does work! And the
quality is fine.
There are two problems we're having now.
1 - From watching the udp fly by, it seems that iconnect does not know
when
we hang up. For example, if I call a voice mail and it starts giving me
its speal and I hang up, iconnect stays connected until the VM hangs up at
its end.
Next, if we try to call out via iconnect from a sip client extension (like
a windows soft phone) all we hear is horrible noise.
Has anyone else had these issues?
Jim
--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz
<[EMAIL PROTECTED]> wrote:
I haven't play around enough to know whether or not the prefix is a
toggle. I will do some experimenting and let you know. Right now I am
prefixing all my calls with .
My experience is that when the carrier's format is G723.1, you can't
hear the incoming voice. When it is in G711 you can. I have made several
calls using G711 and they are acceptable quality. Note that if you
disallow=gsm in the sip.conf file you will get the 488 media errors you
reported earlier.
Below are my config files for sip and the linejack cards:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context=iconnect ; Default for incoming calls
allow=gsm
allow=ulaw
allow=alaw
;register=1813342:[EMAIL PROTECTED]
;register=1202454:[EMAIL PROTECTED]
[iconnecthere]
type=friend
username=
secret=XXX
host=sipauth.deltathree.com
;
; Linux Telephony Interface
;
; Configuration file
;
[interfaces]
mode=dialtone
format=ulaw
echocancel=medium
silencesupression=no
context=local
context=default
txgain=100%
rxgain=100%
device => /dev/phone0
On Tue, 2003-03-11 at 14:28, Jim Archer wrote:
Hi Greg and thanks very much...
A few questions...
First, regarding the prefix, it seemed that this acts as a toggle,
switching from the one codec to the other. But how do I set which me
system uses by default? Or does iconnect always use the high bandwidth
one by default (such that the always switches to the low
bandwidth
one)?
Next, I am still struggling to understand the SIP options and what goes
where. Could you please tell me where the format command goes? Is
this
an option on the channel? I thing the allow goes in sip.conf.
Finally, does this have any impact on the problem where the person
called can not be heard?
Thanks!!!
Jim
--On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz
<[EMAIL PROTECTED]> wrote:
Jim,
I changed my extensions entry for iconnect to:
exten => _1XX,1,Di
[Asterisk-Users] chan_capi version 0.1.0 released
hi alaw (and now also ulaw) folks, version 0.1.0 is out now. thanks to Bicster for beating eicon capi to support ulaw and echo cancellation!! http://www.junghanns.net/chan_capi.html regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705390 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA beginners question
Title: Message
When dialing a port
on my ATA-186 I get:
== Spawn extension
(default, s, 1) exited non-zero on 'SIP/ata1-1-0c77' --
Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
stack -- Executing Dial("SIP/ata1-2-4fc0",
"SIP/ata1-1|30") in new stack -- Called
ata1-1 -- Got SIP response 488 "Not Acceptable Here" back
from 192.168.1.100 == No one is available to answer at this
time
Instead of a ringing
telephone... Both ata1-1 and ata1-2 are registered with
asterisk.
Can anyone tell me
how to get rid of the 488?
Thanks!
Re: R: [Asterisk-Users] Several patches, including recording and music-on-hold
Hi, At 10:30 12-3-2003 +0100, you wrote: (i'm thinking to put up a little website, with a repository of all * patches. tell me if anyone is interested into that) Yes, this is -very- usefull. How about a cvs repository :-) Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Hi Lubo, I appreciate your email to help with this issue, but I don't understand your message. I assume your comment about format=slinear is to use format=slinear in phone.conf instead of format=ulaw. If so, how does this get you g723 to iconnect? Using format=g723.1 doesn't seem to work. Gregg On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote: > Dan, why are you using phonejack with ulaw codec? g723 (format=slinear > only) is working just perfect with phonejack and iconnect :) > > Lubo > > Dan Fernandez wrote: > > I found similar problems. > > > > With my phonejack I can make a call with ulaw with decent quality (I have a > > 64k line). > > > > However, with Messenger I hear a brief horrible noise and thatґs it. > > > > - Original Message - > > From: "Jim Archer" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, March 11, 2003 8:17 PM > > Subject: Re: [Asterisk-Users] iconnect quality? > > > > > > > >>Ok! When I use the prefix and I allow gsm it does work! And the > >>quality is fine. > >> > >>There are two problems we're having now. > >> > >>1 - From watching the udp fly by, it seems that iconnect does not know > > > > when > > > >>we hang up. For example, if I call a voice mail and it starts giving me > >>its speal and I hang up, iconnect stays connected until the VM hangs up at > >>its end. > >> > >>Next, if we try to call out via iconnect from a sip client extension (like > >>a windows soft phone) all we hear is horrible noise. > >> > >>Has anyone else had these issues? > >> > >>Jim > >> > >> > >>--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > >><[EMAIL PROTECTED]> wrote: > >> > >> > >>>I haven't play around enough to know whether or not the prefix is a > >>>toggle. I will do some experimenting and let you know. Right now I am > >>>prefixing all my calls with . > >>> > >>>My experience is that when the carrier's format is G723.1, you can't > >>>hear the incoming voice. When it is in G711 you can. I have made several > >>>calls using G711 and they are acceptable quality. Note that if you > >>>disallow=gsm in the sip.conf file you will get the 488 media errors you > >>>reported earlier. > >>> > >>>Below are my config files for sip and the linejack cards: > >>> > >>>; > >>>; SIP Configuration for Asterisk > >>>; > >>>[general] > >>>port = 5060 ; Port to bind to > >>>bindaddr = 0.0.0.0 ; Address to bind to > >>>context=iconnect ; Default for incoming calls > >>>allow=gsm > >>>allow=ulaw > >>>allow=alaw > >>> > >>>;register=1813342:[EMAIL PROTECTED] > >>>;register=1202454:[EMAIL PROTECTED] > >>> > >>>[iconnecthere] > >>>type=friend > >>>username= > >>>secret=XXX > >>>host=sipauth.deltathree.com > >>> > >>>; > >>>; Linux Telephony Interface > >>>; > >>>; Configuration file > >>>; > >>>[interfaces] > >>> > >>>mode=dialtone > >>>format=ulaw > >>>echocancel=medium > >>>silencesupression=no > >>> > >>>context=local > >>>context=default > >>> > >>>txgain=100% > >>>rxgain=100% > >>>device => /dev/phone0 > >>> > >>> > >>> > >>>On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >>> > Hi Greg and thanks very much... > > A few questions... > > First, regarding the prefix, it seemed that this acts as a toggle, > switching from the one codec to the other. But how do I set which me > system uses by default? Or does iconnect always use the high bandwidth > one by default (such that the always switches to the low > > > > bandwidth > > > one)? > > Next, I am still struggling to understand the SIP options and what goes > where. Could you please tell me where the format command goes? Is > > > > this > > > an option on the channel? I thing the allow goes in sip.conf. > > Finally, does this have any impact on the problem where the person > called can not be heard? > > Thanks!!! > > Jim > > --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > <[EMAIL PROTECTED]> wrote: > > > >Jim, > > > >I changed my extensions entry for iconnect to: > > > >exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED] > > > >and my calls work fine with ulaw. I am calling from a linejack card > >with format=ulaw and SIP with allow=ulaw. > > > >Gregg > > > >On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > > > >>--On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >><[EMAIL PROTECTED]> wrote: > >> > >> > >>>Iconnect uses codecs g723 and g711 that can be configured for each > >>>account (you can change them by the prefix) > >> > >>I tried adding the in front of a number and it reliably > > > > generates > > > >>error "488 invalid media." > >> > >> > >>___ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>
Re: [Asterisk-Users] Cheap source of music on hold music?
Good to know that distinction William X Walsh ([EMAIL PROTECTED]) wrote: > >On Wed, 2003-03-12 at 07:09, Brian Johnson wrote: >> Just got this from my TechTV newsletter: >> >> PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE >> You don't have to download illegally to get great music on the Web. >> http://cgi.techtv.com/memberservices/newsletters?click=18285&release=2560 > >Free to download is one thing, but free to "perform" or rebroadcast >(Which technically music on hold is) is very different. The copyright >holder has to specific grant that right, which is what the "Open Audio" >and similar licenses cover. > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- Brian Johnson This is where my witty signature line would be if I bothered to edit this line :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 and fake ring
> Who is generating this ringback? The ATA or asterisk? What if I call > a non-suping number with a "the number has been changed" recording? > Will I never hear it because audio will never be cut through without > answer supervision? Find out by doing a trace. If you're using callprogress, then you should see a 180 Ringing sent to the ATA when we detect ringing on the FXO. If you're not using call progress, then we should not be sending 180 ringing. You can also use the new "debug channel Zap/1-1" to see if the FXO is ringing. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Several patches, including recording andmusic-on-hold
> Mark, will you merge that ;-) ? > > (i'm thinking to put up a little website, > with a repository of all * patches. tell me > if anyone is interested into that) Usually, on a patch of this size, it takes time to merge because I have to carefully study and understand all the changes that are being made, and be sure that the proposed solution is "the right one" for inclusion in the code base. In this case, the patch is *very good* and I had just a handful of suggestions for the author, which I've sent off-list. Hopefully he will respond back soon with an updated patch and I can apply it. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap source of music on hold music?
On Wed, 2003-03-12 at 07:09, Brian Johnson wrote: > Just got this from my TechTV newsletter: > > PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE > You don't have to download illegally to get great music on the Web. > http://cgi.techtv.com/memberservices/newsletters?click=18285&release=2560 Free to download is one thing, but free to "perform" or rebroadcast (Which technically music on hold is) is very different. The copyright holder has to specific grant that right, which is what the "Open Audio" and similar licenses cover. -- William Walsh <[EMAIL PROTECTED]> Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap source of music on hold music?
Just got this from my TechTV newsletter: PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE You don't have to download illegally to get great music on the Web. http://cgi.techtv.com/memberservices/newsletters?click=18285&release=2560 Karl Putland ([EMAIL PROTECTED]) wrote: > >On Wed, 2003-03-12 at 06:48, Brian Johnson wrote: >> I wondered about using streaming content from the internet .. some is available for >> free and some site allow custom music lists. >> >> I thought it would be cool to access audio from elsewhere in the world and found >> that Reuters (the news source for a lot of media stations) provided streaming audio. >> >> Then I ran out of time and haven't had a chance to get back to it >> >> > >Could use something like streamripper to save the stream to a series of >files in your moh dir. If * recognises new files for moh after start >then it would just continue to play the next file in line. The you'd >need a cron job to sweep up the older files before your disk filled up. > >--Karl > >> >> >> Jim Archer ([EMAIL PROTECTED]) wrote: >> > >> >Hi all... >> > >> >I have been shopping around and noticed that licensed music on hold music >> >can be a bit expensive if you want to assemble a variety of types. Does >> >anyone know of an inexpensive source? >> > >> >Thanks... >> > >> > >> >___ >> >Asterisk-Users mailing list >> >[EMAIL PROTECTED] >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> -- >> Brian Johnson >> >> This is where my witty signature line would be if I bothered to edit this line :) >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- Brian Johnson This is where my witty signature line would be if I bothered to edit this line :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap source of music on hold music?
On Wed, 2003-03-12 at 06:48, Brian Johnson wrote: > I wondered about using streaming content from the internet .. some is available for > free and some site allow custom music lists. > > I thought it would be cool to access audio from elsewhere in the world and found > that Reuters (the news source for a lot of media stations) provided streaming audio. > > Then I ran out of time and haven't had a chance to get back to it > > Could use something like streamripper to save the stream to a series of files in your moh dir. If * recognises new files for moh after start then it would just continue to play the next file in line. The you'd need a cron job to sweep up the older files before your disk filled up. --Karl > > > Jim Archer ([EMAIL PROTECTED]) wrote: > > > >Hi all... > > > >I have been shopping around and noticed that licensed music on hold music > >can be a bit expensive if you want to assemble a variety of types. Does > >anyone know of an inexpensive source? > > > >Thanks... > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Brian Johnson > > This is where my witty signature line would be if I bothered to edit this line :) > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Several patches, including recording and music-on-hold
Will mark merge this into CVS or do I have to apply this to my own code checked out of CVS? Also, which version of CVS code do I patch this against? On Tuesday, March 11, 2003, at 10:00 PM, Fettahlioglu, Mahmut wrote: Hello everyone, I was working on implementing several changes to asterisk. I had described the changes and their reasons in several emails before I started the job. I have completed the modifications some weeks ago; the system was then tested extensively and is now in use in a production environment. It seems fairly stable so far. I believe at least some of the additions can be of general use and would like to submit a patch. The modifications done are: - Call recording. Any channel can be recorded through calling applications on the dial plan, or sending messages through the manager interface. A monitoring resource is created for this functionality. I used Kostya V. Ivanov's call recording patch to get the idea how. Thanks for this Kostya! - An alaw pcm format file driver. I had to record conversations in alaw format. This is based on the ulaw format driver. However, I had to make file writing real-time for call recording to function correctly when channels use silence suppression. To do this, basically the file driver keeps the time recording started, and when new data is to be written to the file, it is written to the correct file position. I have seen that a seek function was added to file drivers after I have done those changes, and I think we might want to change the way file seeking is done currently. However, initially adding the file as is will be the easiest. - Additional information is sent through the manager interface for queue events. - Some bug fixes in chan_agent channel driver (mostly to do with race conditions). - Modification to chan_sip so the default callerid used when sending INVITEs is not hardcoded to "asterisk", but is read from the config file. It defaults to "asterisk". - Modifications to manager to make the tcp connection TCP_NODELAY. I have found we sometimes have delays in tcp message transmit due to the Naple algorithm used in TCP. For this fix to work, both the client (astman, gastman) and the server (asterisk) needs to be in TCP_NODELAY mode. - Asterisk exits if it cannot create a thread. I have observed that once thread creation starts to fail, Asterisk never recovers. So the changes cause asterisk to stop once it detects that. The starter script can then restart asterisk. This behaviour can be disabled by sending -DEXPERIMENTAL to the compiler in the makefile. - Changed how music on hold resource works so that it does not need a zaptel driver for timing any more. Sound quality is good and not choppy without any zaptel driver. I believe this should be about all. The patchfile is attached. Any comments are welcome. Regards, Mahmut --- - - Mahmut Fettahlioglu Software Architect Open Access Pty Ltd PO Box 301 Crows Nest NSW 1585 Phone 02 9978 7009 Fax 02 9978 7099 Email <[EMAIL PROTECTED]> --- - - This email is intended only for the use of the individual or entity named above and may contain information that is confidential and privileged. If you are not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email or telephone 02 9978 7009 and destroy the original message. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap sourc of music on hold music?
On Tue, 2003-03-11 at 23:02, Jim Archer wrote: > Hi all... > > I have been shopping around and noticed that licensed music on hold music > can be a bit expensive if you want to assemble a variety of types. Does > anyone know of an inexpensive source? There are a handful of sites collecting music published under the EFF's Open Audio License that you can download and use for free. Not all of it is good, not all of it is bad. http://www.eff.org/IP/Open_licenses/eff_oal.html This site has a few tracks you can download: http://openmusic.linuxtag.org/ Here is a sample of a collection site of music using the EFF license: http://www.openregistry.org/ http://www.rootrecords.org/ They normally depend on donations. Search for Open Audio, or Open Music, or similar terms in google. -- William Walsh <[EMAIL PROTECTED]> Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap source of music on hold music?
I wondered about using streaming content from the internet .. some is available for free and some site allow custom music lists. I thought it would be cool to access audio from elsewhere in the world and found that Reuters (the news source for a lot of media stations) provided streaming audio. Then I ran out of time and haven't had a chance to get back to it Jim Archer ([EMAIL PROTECTED]) wrote: > >Hi all... > >I have been shopping around and noticed that licensed music on hold music >can be a bit expensive if you want to assemble a variety of types. Does >anyone know of an inexpensive source? > >Thanks... > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Brian Johnson This is where my witty signature line would be if I bothered to edit this line :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 configuration question
pwlib v1_4_10 openh323v1_11_6 Am Mit, 2003-03-12 um 14.19 schrieb Krzysztof Bujak: > Which version of pwlib and openh323 libraries should I have to successfully > compile > chan_h323? > AND not get seg faults? -- ___ Jens-Erik Hansen Global Information Services Steindamm 132, D-24145 Kiel Telefon +49 (0)431 7164191 Telefax +49 (0)431 7164192 E-Mail [EMAIL PROTECTED] ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 configuration question
Which version of pwlib and openh323 libraries should I have to successfully compile chan_h323? AND not get seg faults? Best regards, Krzysztof Bujak - Original Message - From: "Jeremy McNamara" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 10, 2003 8:45 PM Subject: Re: [Asterisk-Users] chan_h323 configuration question > AloowGkRouted=yes > > Jens-E. Hansen wrote: > > >My * registers with gnugk using the alias PBX > > > >[PBX] > >type=h323 > >prefix=100 > >context=voip-h323 > > > >Now if i start a call with 'ohphone -g gatekeeper -u testuser 10010' i > >get 'File chan_h323.c, Line 921 (setup_incoming_call): User 'testuser' > >not found, rejecting call.' > > > >How can i allow unlimited dial in (regardless which user calls) through > >the gatekeeper in routed mode? > > > > > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > === > Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form. > === Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form.
[Asterisk-Users] Stripping SDP body in SIP messages
Hello, * from cvs today, Wed Mar 12 about12:00, don't add SDP 'a' parameter in SIP messages. Any calls between SIP-devices via asterisk. SDP body: v=0 o=root 27982 27982 IN IP4 192.168.50.8 s=session c=IN IP4 192.168.50.8 t=0 0 m=audio 59430 RTP/AVP And log from cisco AS5300 Mar 12 13:42:39 MSK: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 192.168.50.8:5060 Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: sipSPISipIncomingCall Mar 12 13:42:39 MSK: 0x623164C0 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: act_idle_new_message Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: sact_idle_new_message_invite Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: sip_stats_method Mar 12 13:42:39 MSK: sipsdp_is_valid: Error in one of the SDP body fields Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: sipInviteError Mar 12 13:42:39 MSK: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE Mar 12 13:42:39 MSK: CCSIP-SPI-CONTROL: sip_stats_status_code Mar 12 13:42:39 MSK: 0x623164C0 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE) Mar 12 13:42:39 MSK: Sent: SIP/2.0 400 Bad Request - 'Invalid SDP information' cvs version from -D '1 day ago' work properly. v=0 o=root 32049 32049 IN IP4 192.168.50.8 s=session c=IN IP4 192.168.50.8 t=0 0 m=audio 54678 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Mar 12 14:24:59 MSK: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 192.168.50.8:5060 Mar 12 14:24:59 MSK: CCSIP-SPI-CONTROL: sipSPISipIncomingCall Mar 12 14:24:59 MSK: 0x62318850 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) Mar 12 14:24:59 MSK: CCSIP-SPI-CONTROL: act_idle_new_message Mar 12 14:24:59 MSK: CCSIP-SPI-CONTROL: sact_idle_new_message_invite Mar 12 14:24:59 MSK: CCSIP-SPI-CONTROL: sip_stats_method Mar 12 14:24:59 MSK: Using Voice Class Codec, tag=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-186 and fake ring
I am using an ATA-186 connecting to an asterisk SIP gateway. When I dial out through it (via a PRI) to a real number, I notice that I hear a fake ringback tone. For example, if I call my voicemail, which answers without a ring, I still hear a bit of ringback when I call via SIP. In fact, if I called a busy number, I never heard a busy. Just continuous ringback, as if it's just playing me local ringback until it sees answer supervision, at which time it cuts the call through. I alleviated this by adding a line: exten=_XX,3,Busy so now it goes to busy when the number I call is busy, but, actually, I still hear a ringback tone first, and then it goes to busy. Who is generating this ringback? The ATA or asterisk? What if I call a non-suping number with a "the number has been changed" recording? Will I never hear it because audio will never be cut through without answer supervision? The relevant lines from my extensions.conf: ; match any US, and strip leading 1 off exten=_1XX,1,StripMSD,1 ; dial outbound on trunk group 1 exten=_XX,2,Dial,Tor/g1/BYEXTENSION ; if we don't put this in, we'll hear ringback forever on a busy number exten=_XX,3,Busy Thanks for putting up with this relatively green asterisk user... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SQL
Hello everyone, I would like to have soneone who knows how to use SQL method to update asterisk's conf files, without disrupting the ongoing calls. Can someone give me some sample about using SQL to update conf files. Thanks George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Several patches, including recording and music-on-hold
Very nice work. I wonder if mark will include that into the main cvs tree. There're many patches till now that're interesting (like that, or pauline i4l patches), but the risk is to have a lot of confusion. Mark, will you merge that ;-) ? (i'm thinking to put up a little website, with a repository of all * patches. tell me if anyone is interested into that) Matteo Brancaleoni [EMAIL PROTECTED] Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it > -Messaggio originale- > Da: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Per conto di > Fettahlioglu, Mahmut > Inviato: mercoledì 12 marzo 2003 4.01 > A: '[EMAIL PROTECTED]' > Cc: '[EMAIL PROTECTED]'; '[EMAIL PROTECTED]' > Oggetto: [Asterisk-Users] Several patches, including > recording and music-on-hold > > > Hello everyone, > > I was working on implementing several changes to asterisk. I > had described the changes and their reasons in several emails > before I started the job. I have completed the modifications > some weeks ago; the system was then tested extensively and is > now in use in a production environment. It seems fairly stable so far. > > I believe at least some of the additions can be of general > use and would like to submit a patch. The modifications done are: > > - Call recording. Any channel can be recorded through calling > applications on the dial plan, or sending messages through > the manager interface. A monitoring resource is created for > this functionality. I used Kostya V. Ivanov's call recording > patch to get the idea how. Thanks for this Kostya! > - An alaw pcm format file driver. I had to record > conversations in alaw format. This is based on the ulaw > format driver. However, I had to make file writing real-time > for call recording to function correctly when channels use > silence suppression. To do this, basically the file driver > keeps the time recording started, and when new data is to be > written to the file, it is written to the correct file > position. I have seen that a seek function was added to file > drivers after I have done those changes, and I think we might > want to change the way file seeking is done currently. > However, initially adding the file as is will be the easiest. > - Additional information is sent through the manager > interface for queue events. > - Some bug fixes in chan_agent channel driver (mostly to do > with race conditions). > - Modification to chan_sip so the default callerid used when > sending INVITEs is not hardcoded to "asterisk", but is read > from the config file. It defaults to "asterisk". > - Modifications to manager to make the tcp connection > TCP_NODELAY. I have found we sometimes have delays in tcp > message transmit due to the Naple algorithm used in TCP. For > this fix to work, both the client (astman, > gastman) and the server (asterisk) needs to be in TCP_NODELAY mode. > - Asterisk exits if it cannot create a thread. I have > observed that once thread creation starts to fail, Asterisk > never recovers. So the changes cause asterisk to stop once it > detects that. The starter script can then restart asterisk. > This behaviour can be disabled by sending -DEXPERIMENTAL to > the compiler in the makefile. > - Changed how music on hold resource works so that it does > not need a zaptel driver for timing any more. Sound quality > is good and not choppy without any zaptel driver. > > I believe this should be about all. The patchfile is > attached. Any comments are welcome. > > Regards, > > Mahmut > > -- > -- > - > Mahmut Fettahlioglu > Software Architect > > Open Access Pty Ltd > PO Box 301 > Crows Nest NSW 1585 > > Phone 02 9978 7009 > Fax 02 9978 7099 > Email <[EMAIL PROTECTED]> > -- > -- > - > This email is intended only for the use of the individual or > entity named above and may contain information that is > confidential and privileged. If you are not the intended > recipient, you are hereby notified that any dissemination, > distribution or copying of this email is strictly prohibited. > If you have received this email in error, please notify us > immediately by return email or telephone > 02 9978 7009 and destroy the original message. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
