Re: [asterisk-users] How to connect asterisk PBX to PSTN
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jigar Joshi Sent: Tuesday, October 19, 2010 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to connect asterisk PBX to PSTN I will have a closer look at this book My Question is : Is it possible with asterisk to resolve all the code to one extension and with the extension no. For example person one calls it and enters code 1 person two calls and enters code 2 both the call should received by a single extension say 1001. and there I should be able to differenciate both the calls using code entered. in the example: both the call will be given to extension 1001 and at 1001 there will be an app running that will make this into two calls.i mean each packet contains the code entered. I hope this answer would be helpful . Thanks. snip #1. Steve (as usual) is correct #2. We eat this kind of simple dialplanning for lunch #3. After you read the book, ask it again if needed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available
On 10-10-18 11:01 PM, Barry Miller wrote: On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote: On 10-10-18 07:54 PM, Asterisk Development Team wrote: For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4 Apologies, this link should be: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 -- The Asterisk Development Team Is it worth mentioning somewhere (ChangeLog? This list?) that all the asterisk-core-sounds tarballs were updated today? It would remind someone [me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a 'make sounds' before stopping asterisk for the install. My test system is on a slow link, and waiting for the tarball downloads in the middle of installing is frustrating. If you deselect the sounds from menuselect then you don't have to wait for them to download, and you can update them at your convenience later. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
This would be overkill, but I thought I would mention these products in case someone was looking for a more robust/industrial automation solution that could be integrated with the linux command line and asterisk. I do use asterisk as a remote control device for a few non-critical inputs (air supply valves) in our production plant. I also use asterisk for remotely checking status on few IO points. Opto 22 produces the SNAP PAC System a line of IO control brains. Some of these devices/brains can run a control program independently, other serve as remote IO points for a controlling brain or a program running on your linux server. Most of the software is free with the exception of a few programs that are used in a large scale deployment situation. Cheapest non-controlling brain http://www.opto22.com/site/pr_details.aspx?cid=4item=SNAP-PAC-EB2 SDK for linux development (free). The SDK contains a program called eioctl that you can compile for command line control of IO points. http://www.opto22.com/site/pr_details.aspx?cid=4item=SNAP-PAC-EB2 http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 On Mon, Oct 18, 2010 at 8:09 PM, Bryant Zimmerman brya...@zktech.comwrote: I would look at x10 triggered switches. There are some command line tools you could call from an IVR. I did a lot of x10 development on windows back in the day. I have seen some things for linux as well. http://www.heyu.org/ Bryant -- *From*: C F shma...@gmail.com *Sent*: Monday, October 18, 2010 7:55 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk to switch on electric heaters remotely? Ah Sandman http://sandman.com use a relay that goes onto an fxs port, call that fxs port and you have a connection. Since that only work momentary you will need a flip flop relay, the advantage is that by calling it again you can turn it off. Ring relay: http://sandman.com/wizard.html#UniversalRingRelay flip flop relay: http://altronix.com/index.php?pid=2model_num=RBR1224 On Mon, Oct 18, 2010 at 7:09 AM, Gilles codecompl...@free.fr wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marvin Horst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integrate Intertel Axxess with Asterisk
How did the setup work as far as extensions on the Inter-Tel system contacting extensions on the asterisk system? On Thu, Oct 14, 2010 at 9:56 AM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: We have it integrated, but differently; we have 2 T1 voice lines, and a 4-port Sangoma card. The T1 lines run into the Asterisk system, and then the other two ports run into the Inter-Tel. If the Asterisk system has directives that match a call, it does something with it. Otherwise, it passes it back out to the Inter-Tel. This has worked surprisingly well, and makes it easy to transition away from the Inter-Tel equipment at our own speed. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *marvin horst *Sent:* Wednesday, October 06, 2010 5:00 PM *To:* asterisk-users *Subject:* [asterisk-users] integrate Intertel Axxess with Asterisk Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -- Marvin Horst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:98765...@xxx.xxx.xxx.xx8:5060 Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv - --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. --- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:1234...@yyy.yyy.yyy.yyy Content-Length: 0 --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Please help. Thank you. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
Hi Cassius On the iax conf for both box change the secret parameter with remotesecret. This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X Regards 2010/10/18 Cassius Smith cass...@cassius.org I'm having trouble getting an IAX2 connection between a couple of servers. I can make calls from server B to server A, but when I call from Server A to server B, I get No authority found. If I remove serverA's password on ServerB's iax.conf, calls will go through as UNAUTHENTICATED. On ServerA I am running Asterisk 1.6.2.9 On ServerB I'm running 1.6.2.13 Any hints for me? The registrations in both directions seem to work fine when I do an iax2 reload from the CLI. config file snips shown below. Thanks Cassius Smith = On server B, I have the following: [general] register = serverB:longsecretpasswo...@servera_ip [serverA] type=friend host=dynamic auth=md5 secret=longsecretpassword1 context=no911 [serverB] type=friend host=dynamic auth=md5 secret=longsecretpassword2 ; if I remove this, calls go through as UNAUTHENTICATED context=no911 On server A, I have the following: [general] register = serverA:longsecretpasswo...@serverb_ip [serverB] type=friend host=dynamic auth=md5 secret=longsecretpassword2 context=no911 [cary] type=friend host=dynamic auth=md5 secret=longsecretpassword1 context=no911 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi vmware query
Greeting list, I hope this isn't a lazy question. I have been running TDM400P and TDM410P cards in Dell PowerEdge Servers for a few years now. We are moving from physical servers to VMWARE servers. What opportunities should I expect moving these cards into the new machines? Or should I leave the existing machines intact and use IAX to get to the DAHDI lines from the VMWARE servers? Thanks for your input Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote: Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite take a look at your canreinvite option. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect asterisk PBX to PSTN
Hi, I think as more than one already replied this is feasible, but I would like to underline that it's better to have a look to that book at least to know how to ask and provide more details to this mailing list by using a more VoIP/Asterisk standard language. BTW I think you would like to implement a queue manager, that is already available on Asterisk. Of course, this is what I understood from your question. On Tue, Oct 19, 2010 at 4:01 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jigar Joshi *Sent:* Tuesday, October 19, 2010 1:28 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to connect asterisk PBX to PSTN I will have a closer look at this book My Question is : Is it possible with asterisk to resolve all the code to one extension and with the extension no. For example person one calls it and enters code 1 person two calls and enters code 2 both the call should received by a single extension say 1001. and there I should be able to differenciate both the calls using code entered. in the example: both the call will be given to extension 1001 and at 1001 there will be an app running that will make this into two calls.i mean each packet contains the code entered. I hope this answer would be helpful . Thanks. snip #1. Steve (as usual) is correct #2. We eat this kind of “simple dialplanning” for lunch #3. After you read the book, ask it again if needed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
It's set to yes for this peer. also t38pt_udptl is set to yes. :( On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote: Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite take a look at your canreinvite option. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect asterisk PBX to PSTN
Thanks All, I will look into book for sure. On Tue, Oct 19, 2010 at 8:50 PM, Marino Punturieri map...@gmail.com wrote: Hi, I think as more than one already replied this is feasible, but I would like to underline that it's better to have a look to that book at least to know how to ask and provide more details to this mailing list by using a more VoIP/Asterisk standard language. BTW I think you would like to implement a queue manager, that is already available on Asterisk. Of course, this is what I understood from your question. On Tue, Oct 19, 2010 at 4:01 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jigar Joshi *Sent:* Tuesday, October 19, 2010 1:28 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to connect asterisk PBX to PSTN I will have a closer look at this book My Question is : Is it possible with asterisk to resolve all the code to one extension and with the extension no. For example person one calls it and enters code 1 person two calls and enters code 2 both the call should received by a single extension say 1001. and there I should be able to differenciate both the calls using code entered. in the example: both the call will be given to extension 1001 and at 1001 there will be an app running that will make this into two calls.i mean each packet contains the code entered. I hope this answer would be helpful . Thanks. snip #1. Steve (as usual) is correct #2. We eat this kind of “simple dialplanning” for lunch #3. After you read the book, ask it again if needed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote: It's set to yes for this peer. also t38pt_udptl is set to yes. :( You don't say anything about what you're trying to send / receive against. Here's how you should troubleshoot: * start with a 'real fax machine' if you have one, on an analog line if you have one. If you can't receive / send with that against your target, blame your target. * move to audio-pass through fax on asterisk. No T.38. If that works. * add in T.38 You will learn things in that process and be able to tell at what layer your troubles are happening. It could be coincidental that things give up during the reinvite. It could actually be giving up for noise on the line, packet drops, etc. At the very least, start recording the call. You'll at least be able to hear up to the re-invite. Definitely record the audio passthrough attempt and listen back to it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integrate Intertel Axxess with Asterisk
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst fivehor...@gmail.com wrote: How did the setup work as far as extensions on the Inter-Tel system contacting extensions on the asterisk system? It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel worked. Still had to watch out for line error counters, and still had to reboot it daily (Windows + Inter-Tel equals unstable). When sending calls into Inter-Tel, the other side, probably asterisk, masquerades as telco sending in call as PRI. In the other direction, you configure the lines as OPX, or off-premise extensions. Just make the extensions match for each line on the PRI and set your dialplan so you keep things making sense. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
We don't have an ATA and fax machine. The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. As I showed in the logs, the remote carrier sends a proper T.38 reINVITE, but our Asterisk doesn't accept, despite the fact that this provider is defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the only question is (as far as we understand) is why in this scenario, the T.38 is rejected. Here are the logs (sip debug is open) again, since we get the reINVITE: --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060 Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv - --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. --- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy Content-Length: 0 --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Thanks. Michael On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote: It's set to yes for this peer. also t38pt_udptl is set to yes. :( You don't say anything about what you're trying to send / receive against. Here's how you should troubleshoot: * start with a 'real fax machine' if you have one, on an analog line if you have one. If you can't receive / send with that against your target, blame your target. * move to audio-pass through fax on asterisk. No T.38. If that works. * add in T.38 You will learn things in that process and be able to tell at what layer your troubles are happening. It could be coincidental that things give up during the reinvite. It could actually be giving up for noise on the line, packet drops, etc. At the very least, start recording the call. You'll at least be able to hear up to the re-invite. Definitely record the audio passthrough attempt and listen back to it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA - it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
On Tue, Oct 19, 2010 at 10:44 AM, Andrea Sannucci asannu...@gmail.com wrote: On the iax conf for both box change the secret parameter with remotesecret. This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X This is incorrect, chan_iax.so does not have such a parameter. However, chan_sip.so does, and it is documented. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
On Mon, 18 Oct 2010 13:09:50 +0200, Gilles codecompl...@free.fr wrote: I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. Thanks everyone for the great feedback. Following Steve Edward's advice, I won't automate the process and will only switch the heaters on manually by dialing into the IVR or sending an e-mail. It's an individual office to which I have the only key, so there's no risk of the heaters eg. being covered with clothes. I'll check the following sites X10 and Arduino, and the other resources mentionned above. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
Hi Paul, I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with username and password. Only when i changed secret with remotesecret the connection work. Maybe you can try the same configuration to confirm this behaviour Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question voip.quest...@gmail.com wrote: The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. Asterisk can handle audio passthrough faxing. I'm talking audio faxing over SIP. You compile against this thing called SpanDSP, and then asterisk squawks audio tones over the line. It's amazing. Until you've tried it, you don't know whether it could work. I'm under the assumption that you'd rather have faxing at all than faxing over T.38. The world is littered with broken T.38 implementations. Just because it's a standard, doesn't mean people follow it. Ever heard of HTML? Which browsers follow it to the letter? You seem to have never successfully exchanged a fax with your target, so I don't know why you think the far end isn't broken. Try turning off t38 and see what happens. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with username and password. Only when i changed secret with remotesecret the connection work. I would enable iax debugs and confirm you calls you being authenticated, and not using a guest account. As I mentioned, 'remotesecret' is not an option for chan_iax2.so -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith cass...@cassius.org wrote: Any hints for me? Server Ottawa (192.168.1.190) register = Ottawa:ottawaisc...@192.168.1.196 [Toronto] type=peer host=dynamic username=Toronto secret=TorontoIsFine Server Toronto (192.168.1.196) register = Toronto:torontoisf...@192.168.1.190 [Ottawa] type=peer host=dynamic username=Ottawa secret=OttawaIsCool -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. I spent a few hours on Google and saw many similar posts, but no actual valuable answer. Weird... On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas da...@debsinc.com wrote: From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA – it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
I think the generic throw away gmail address will keep many people from answering... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Oct 19, 2010 at 2:01 PM, VoIP Question voip.quest...@gmail.com wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. I spent a few hours on Google and saw many similar posts, but no actual valuable answer. Weird... On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas da...@debsinc.com wrote: From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA – it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question voip.quest...@gmail.com wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I don't know. I'm not using FFA, and I'm doing more channels simultaneously than I want to disclose. It's called app_fax, and it's been built into the 1.6 series for quite some time now. You seem pretty hell bent against spending any money. Not for FFA, not for commercial digium support, not for an analog copper line, not for an ATA. So try app_fax. It's free. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
Hi Cassius, it may be slightly offsubject but i did connect two 1.6 asterisk boxes, and the only issue i had is these two statements missing: calltokenoptional=209.16.236.73/255.255.255.0 requirecalltoken=no hope it helps! On Tue, Oct 19, 2010 at 12:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith cass...@cassius.org wrote: Any hints for me? Server Ottawa (192.168.1.190) register = Ottawa:ottawaisc...@192.168.1.196ottawa%3aottawaisc...@192.168.1.196 [Toronto] type=peer host=dynamic username=Toronto secret=TorontoIsFine Server Toronto (192.168.1.196) register = Toronto:torontoisf...@192.168.1.190toronto%3atorontoisf...@192.168.1.190 [Ottawa] type=peer host=dynamic username=Ottawa secret=OttawaIsCool -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi vmware query
On 10-10-19 10:46 AM, Danny Nicholas wrote: Greeting list, I hope this isn’t a “lazy” question. I have been running TDM400P and TDM410P cards in Dell PowerEdge Servers for a few years now. We are moving from physical servers to VMWARE servers. What opportunities should I expect moving these cards into the new machines? Or should I leave the existing machines intact and use IAX to get to the DAHDI lines from the VMWARE servers? Only the host system will be able to see the cards, so regardless you will need to use IAX or SIP to access the Asterisk instance that is hosting the hardware. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 channels real time monitoring
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute dahdi show channels but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ??? Thanks a lot Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 channels real time monitoring
- Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute dahdi show channels but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ??? I'm not sure what is considered the 'best way' but you could script something like: asterisk -rx 'core show channels' | grep DAHDI | regex stuff/etc/pass to rrdtool --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 works one direction, but not the other...
Hi Paul, maybe there is some think wrong on iax. if I set remotesecret on IAX2 extension the call from Server A to Server B work but not authenticated and host is set to dynamic (normaly if is a IP authentication on host parametre I put the IP) If I set secret on two box and both are registered without errors, when i call from box A or from box B to box A, always I receive this error No auhthority found. Thi behaviour only happens if the Asterisk version onTwo box is 1.6.2.13. If Asterisk version on Box A is 1.4.X and Asterisk version on Box B is 1.6.2.13 (with the same configuration) work fine. Why? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/19/2010 10:48 AM, VoIP Question wrote: We don't have an ATA and fax machine. The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. As I showed in the logs, the remote carrier sends a proper T.38 reINVITE, but our Asterisk doesn't accept, despite the fact that this provider is defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the only question is (as far as we understand) is why in this scenario, the T.38 is rejected. Here are the logs (sip debug is open) again, since we get the reINVITE: Is the SendFAX application running when this re-INVITE is received? You didn't actually include a complete console log, only the SIP traces, so we can't see what was happening in Asterisk at this time. You also didn't include an Asterisk version number (or any version numbers), so we can't be sure whether the problem you are seeing is a known resolved one or not. Please post the complete console log, with all logger levels enabled, so we can see the entire timeline of the call up to the failure occurring. If the re-INVITE arrives before SendFAX is started, there won't be an application to respond to it, and chan_sip will (rightly) assume that T.38 cannot be used on this channel so it will respond with a 488. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/19/2010 12:01 PM, VoIP Question wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. When you are asking for free help on a mailing list, patience is a virtue :-) You posted your question approximately four hours ago. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Fair enough Kevin :-) It's just that your documentation for this product is so limited that without extensive search online and the assistance of others, it would have been impossible for us to make any progress and we haven't reached the ReceiveFax part yet ;) Anyway, specifically, we installed Asterisk 1.6.2.11. As far as we know/understand, the SendFax application is running. This is the full log of the call, until it's rejected for the first time. The remote switch resends the INVITEs a few more times, but it's all the same, so I didn't include it: sip*CLI -- Attempting call on Local/12345...@outgoing for s...@outboundfax:1 (Retry 1) sip*CLI -- Executing [12345...@outgoing:1] Dial(Local/12345...@outgoing-2c36;2, SIP/12345...@main,50,tTr) in new stack sip*CLI == Using SIP RTP CoS mark 5 sip*CLI == Using SIP VRTP CoS mark 6 sip*CLI == Using UDPTL CoS mark 5 sip*CLI Audio is at yyy.yyy.yyy.yyy port 10714 sip*CLI Adding codec 0x100 (g729) to SDP sip*CLI Adding codec 0x2 (gsm) to SDP sip*CLI Adding non-codec 0x1 (telephone-event) to SDP sip*CLI Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060: INVITE sip:12345...@xxx.xxx.xxx.xx8 SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5b6bc617;rport Max-Forwards: 70 From: Fax sip:98765...@yyy.yyy.yyy.yyy;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8 Contact: sip:98765...@yyy.yyy.yyy.yyy Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE User-Agent: PBX Date: Tue, 19 Oct 2010 16:41:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 697508180 697508180 IN IP4 yyy.yyy.yyy.yyy s=Asterisk PBX 1.6.2.11 c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 10714 RTP/AVP 18 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sip*CLI -- Called 12345...@main sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Content-Length: 0 - --- (7 headers 0 lines) --- sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Contact: sip:12345...@xxx.xxx.xxx.xx8:5060 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=audio 6256 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x102 (gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port xxx.xxx.xxx.xx7:6256 sip*CLI -- SIP/main-002a is making progress passing it to Local/12345...@outgoing-2c36;2 sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:12345...@xxx.xxx.xxx.xx8:5060 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=audio 6256 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - --- (15 headers 12 lines) --- list_route: hop: sip:12345...@xxx.xxx.xxx.xx8:5060 set_destination: Parsing sip:12345...@xxx.xxx.xxx.xx8:5060 for address/port to send to set_destination: set destination to xxx.xxx.xxx.xx8, port 5060 Transmitting (no NAT) to
Re: [asterisk-users] integrate Intertel Axxess with Asterisk
From: marvin horst [mailto:fivehor...@gmail.com] Sent: Tuesday, October 19, 2010 10:23 AM To: Justin Sherrill; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] integrate Intertel Axxess with Asterisk How did the setup work as far as extensions on the Inter-Tel system contacting extensions on the asterisk system? I would delete a user's station, and then create a phantom extension with their extension in the Axxcess system, that when dialed, would forward to the Asterisk system's DID for that user. Asterisk would pick it up on the way out and connect it, so it didn't end up eating lines in our T1. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distortion and block on analog lines
Hi listers! Have a problem with distortion in some analog lines. When some call comes in from PSTN the sound is really distorte, nothing can be understanded, but Internal calls work ok. Funny thing is that when I start/stop asterisk,dahdi, and wanrouter services eveything goes fine again. This is happening every week or so. I'm using asterisk 1.4.36, dahdi linux 2.2.0.2 and wanpipe 3.4.9 ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-3.4.9.tgzstable version, as you can guess I'm using Sangoma cards, specifically A400BRMDE Sometimes it also happens that the lines block, so I'm unable to make outbound calls using those lines. both problems solve after services restart any ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available
On Tue, Oct 19, 2010 at 09:59:34AM -0400, Leif Madsen wrote: On 10-10-18 11:01 PM, Barry Miller wrote: On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote: On 10-10-18 07:54 PM, Asterisk Development Team wrote: For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4 Apologies, this link should be: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 -- The Asterisk Development Team Is it worth mentioning somewhere (ChangeLog? This list?) that all the asterisk-core-sounds tarballs were updated today? It would remind someone [me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a 'make sounds' before stopping asterisk for the install. My test system is on a slow link, and waiting for the tarball downloads in the middle of installing is frustrating. If you deselect the sounds from menuselect then you don't have to wait for them to download, and you can update them at your convenience later. Thanks. I would not have thought of that. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked calls drop asterisk-1.4.22.1
Hi We are facing a problem for orphaned parked calls, we have the following config: asterisk -1.4.22.1 dahdi-linux-complete-2.2.0.2+2.2.0 and when we get an incoming call and after it gets parked, after some set time (here its 2 min), it goes back to the operator, but the problem is that randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the extension number of the operator) and we get the error as Unable to create the channel of type SIP (cause code 20) and then the call drops, we even tried asterisk-1.4.23.2, but in that version we were having problems with paging/intercom using the phones. [Oct 19 11:55:28] VERBOSE[2996] logger.c: == Timeout for SIP/5060-b781fe80 parked on 71. Returning to park-dial,SIP/5060,1 [Oct 19 11:55:28] VERBOSE[14641] logger.c: -- Executing [SIP/5...@park-dial:1] Dial(SIP/5060-b781fe80, SIP/5060|30|t) in new stack [Oct 19 11:55:28] WARNING[14641] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Oct 19 11:55:28] VERBOSE[14641] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Oct 19 11:55:28] VERBOSE[14641] logger.c: == Auto fallthrough, channel 'SIP/5060-b781fe80' status is 'CHANUNAVAIL' We also have the option of Page/Intercom through the phones that auto answer. Can any one share any ideas or opinions? Thank you, Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi_genconf
Hi , Please, I am trying to understand the hardware installation on asterisk and I have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and then I run the dahdi_genconf , It create the dahdi_channels and system.conf. Therefore, it is created with a kind of signalling that is not used in my country. Can I edit it? regards! Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf
On 10/19/10 10:03 PM, Flavio Miranda wrote: Please, I am trying to understand the hardware installation on asterisk and I have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and then I run the dahdi_genconf , It create the dahdi_channels and system.conf. Therefore, it is created with a kind of signalling that is not used in my country. Can I edit it? Yes..dahdi_genconf just makes the best guess it can based on the hardware that is loaded. You should feel free to edit those files. Keep in mind that if you rerun dahdi_genconf your changes could be overwritten. dahdi_genconf can be thought of as a tool to get you started with your configuration. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_genconf
Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Just one more question, what it means the RED under alarms when I type dahdi show status. It should be OK? Thanks for your guidance! Date: Tue, 19 Oct 2010 22:38:25 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi_genconf On 10/19/10 10:03 PM, Flavio Miranda wrote: Please, I am trying to understand the hardware installation on asterisk and I have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and then I run the dahdi_genconf , It create the dahdi_channels and system.conf. Therefore, it is created with a kind of signalling that is not used in my country. Can I edit it? Yes..dahdi_genconf just makes the best guess it can based on the hardware that is loaded. You should feel free to edit those files. Keep in mind that if you rerun dahdi_genconf your changes could be overwritten. dahdi_genconf can be thought of as a tool to get you started with your configuration. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
