Re: [asterisk-users] SIP over SSL TCP or SRTP?
22 jun 2012 kl. 21:59 skrev Bruce B: Thanks. Want to secure everything and anything possible. 1- Can both SIP over TLS and SRTP work in conjunction to each other? Yes. As Kevin said, SIP over TLS only secures the signalling. And it secures it hop-by-hop so every server in the middle can access the content. The signalling should be hidden from other Wifi users, even if it's not hidden all the way between caller and callee. In the signalling you specify how to exchange the actual media. To have secure signalling with TLS doesn't necessarily mean that them media (audio/video/text) is secured. The media is secured with Secure RTP or SRTP, which means that every audio packet is encrypted. 2- Is SIP over TLS a package or added on module that can be installed from Digium Asterisk repository? It's part of the current Asterisk SIP stack, but still marked as experimental as it has a number of known issues that needs to be fixed in order to put this in production use in larger sites and networks. You will have to test it to make sure it works for you. Experimental status means that the configuration options may change in a coming release without being backwards compatible. The TLS part has been experimental in many releases without anyone putting any funding towards fixing it. I guess serious use of TLS is done not with Asterisk but with a SIP proxy like Kamailio or OpenSIPS in front of Asterisk. 3- SRTP takes care of the RTP and makes it secure so that MITM type sniffing is not possible? Yes, provided that the media encryption key exchange is secured. Today, the key exchange is done in SIP messaging, which is why you also want SIP over TLS. Regards, /Olle Regards, On Fri, Jun 22, 2012 at 2:39 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/22/2012 12:56 PM, Bruce B wrote: Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. SIP over TLS (what used to be called SSL) is what secures the SIP signaling. SRTP is for securing media streams. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- _ -- 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] low success rate for ReceiveFax
I have set the clock source from MASTER to NORMAL in the sangoma card settings, and I'm still getting 3RD_T2_TIMEOUT error codes: -- Channel 'DAHDI/i1/-4' FAX session '3' is complete, result: 'FAILED' (FAX_FAILURE_PROTOCOL_ERROR), error: '3RD_T2_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' -- Auto fallthrough, channel 'DAHDI/i1/-4' status is 'UNKNOWN' -- Executing [h@fax-rx:1] AGI(DAHDI/i1/-4, receivefax.php,65126150,fax-65126150-1340867263-rx.tif,FAILED,) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/receivefax.php -- DAHDI/i1/-4AGI Script receivefax.php completed, returning 0 -- Executing [h@fax-rx:2] NoOp(DAHDI/i1/-4, FAXOPT(ecm) : yes) in new stack -- Executing [h@fax-rx:3] NoOp(DAHDI/i1/-4, FAXOPT(filename) : /var/spool/asterisk/fax/fax-65126150-1340867263-rx.tif) in new stack -- Executing [h@fax-rx:4] NoOp(DAHDI/i1/-4, FAXOPT(headerinfo) : MY FAXBACK RX) in new stack -- Executing [h@fax-rx:5] NoOp(DAHDI/i1/-4, FAXOPT(localstationid) : 1234567890) in new stack -- Executing [h@fax-rx:6] NoOp(DAHDI/i1/-4, FAXOPT(maxrate) : 14400) in new stack -- Executing [h@fax-rx:7] NoOp(DAHDI/i1/-4, FAXOPT(minrate) : 2400) in new stack -- Executing [h@fax-rx:8] NoOp(DAHDI/i1/-4, FAXOPT(pages) : 0) in new stack -- Executing [h@fax-rx:9] NoOp(DAHDI/i1/-4, FAXOPT(rate) : 2400) in new stack -- Executing [h@fax-rx:10] NoOp(DAHDI/i1/-4, FAXOPT(remotestationid) : ) in new stack -- Executing [h@fax-rx:11] NoOp(DAHDI/i1/-4, FAXOPT(resolution) : unknown) in new stack -- Executing [h@fax-rx:12] NoOp(DAHDI/i1/-4, FAXOPT(status) : FAILED) in new stack -- Executing [h@fax-rx:13] NoOp(DAHDI/i1/-4, FAXOPT(statusstr) : FAX_FAILURE_PROTOCOL_ERROR) in new stack -- Executing [h@fax-rx:14] NoOp(DAHDI/i1/-4, FAXOPT(error) : 3RD_T2_TIMEOUT) in new stack Here are the fax settings: FAX For Asterisk Settings: ECM: Enabled Status Events: Off Minimum Bit Rate: 2400 Maximum Bit Rate: 14400 Modem Modulations Allowed: V17,V27,V29 FAX Technology Modules: DIGIUM (Digium FAX Driver) Settings: Maximum T.38 Packet Delay: 800 T.38 Session Packet Capture: On G.711 Session Audio Capture: On And here's the Wanpipe Config: CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 4 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO TE_RX_SLEVEL= 430 HW_RJ45_PORT_MAP = DEFAULT LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END= NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF= YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL# OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO# NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION= NO# NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO# NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE= NO# NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN= 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN= 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 8 On Tue, Jun 26, 2012 at 10:22 PM, Steve Underwood ste...@coppice.org wrote: On 06/26/2012 11:47 AM, Roi Stork wrote: In what way was my question not meaningful? Not enough details? Enoughj? You didn't give any. Here's our current receive fax route: sender fax machine - telco - E1 line - sangoma card - asterisk We're currently using free fax for asterisk. This constitutes a meaningful question. I have read that fax over voip is not reliable, but is it the same case for faxes going through dahdi channels? It's strange because I previously tested using another
Re: [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
On 27.06.2012 18:46, Michelle Konzack wrote: Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus? Which PCI-ID is that? Armin -- _ -- 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] 2GB Elastix memory limit
I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory as the server is multipurpose Thanks Sam -- _ -- 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] .lock file issue
I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] .lock file issue
- Original Message - I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office). This was quite common in some old releases, at least for me. At one point I wrote a quick script that ran via cron to remove those lock files once per minute. If this isn't a new bug, it could also be a full filesystem, or maybe the system lost power during an event where a lock was created but not removed? --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] 2GB Elastix memory limit
Hi... Since you use PAE kernel the server must be a 32bit machine i' m guessing this is a compiling issue... and i' m not sure how you can get past this issue... On Thu, Jun 28, 2012 at 11:58 AM, resea...@businesstz.com wrote: I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory as the server is multipurpose Thanks Sam -- _ -- 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 -- Best Regards. Yasin SULUHAN Asterisk Telephony Infrastructure Consultant Contact Information Mobile: +90 535 656 35 55 Blog: http://planetvoip.wordpress.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] SendFAX timestamp
On 06/27/2012 09:30 PM, David Cunningham wrote: Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? No, SendFAX (in res_fax) doesn't currently offer the ability to do what you are asking about. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] 2GB Elastix memory limit
El 28/06/12 03:58, resea...@businesstz.com escribió: I have sevaral elastix installed but all of them show the physical memory is 2GB while the server has 4GB and some has 8GB. I've upgraded to PAE kernel but yet i cant see mem beyond 2GB. How can i configure the centos kernel to use more memory as the server is multipurpose I think you should use the Elastix mailing lists for this question. But you should try using the 64-bit Elastix instead. -- _ -- 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] .lock file issue
could also be a full filesystem, or maybe the system lost power during an event where a lock was created but not removed? /var has 110GB free System is on a new UPS. And doing a Google shows hits from 2011 and earlier. Nothing noted this year. Thanks for the input. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] PRI trunk between Asterisk servers does not work.
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] PRI trunk between Asterisk servers does not work.
On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 Hello Ernie, Could you post the dahdi/system.conf from both voip1 and voip3 servers? I suspect that you have not correctly defined the data channel (dchan setup should be in system.conf and not in chan_dahdi.conf, where I see a not necessarily dchannel configuration) HTH, Ioan -- _ -- 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] Forcing SIP trunk matching order?
Hi James On 29/06/2012, at 6:19 AM, James Lamanna wrote: Hi, I have a bunch of different customers on an Asterisk Box (the PBX). This Asterisk Box is behind another Asterisk box that provides a PSTN connection. Up to this point I've been using IAX between the 2 Asterisk boxes, but I would like to use SIP instead. After doing some testing I have the following issue. If customer A calls customer B, but the call goes out through the PSTN and comes back in, the call is rejected at the PBX because it wants authentication. This raises the question of how calls come in from the PSTN in the first place. I am guessing you route them out in order to bill them? I am guessing you have more than one SIP trunk between the two boxes for different purposes and what you are saying is the authentication is falling to the lowest common denominator. In this situation you could separately register two lines, raise your level of security ( insecure = no option), you need to make sure there is an incoming context that matches the supplied user name otherwise it will fall back to the ip address and whatever context that matches I can guess that this must be because it matches the To address to the friend SIP trunk that provides registration for the phone. (All phones are setup as type=friend, host=dynamic). Is there any way to force matching against the peer SIP trunk to PSTN so as to not require authentication for calls coming in from the PSTN server? Using permit and deny directives for the ip address should help here Thanks. -- James -- _ -- 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] PRI trunk between Asterisk servers does not work.
On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). -- _ -- 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
