Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that was not answered and I did not see any more information. The dumpchan of DADHI/23-1 did not happen as that is in a macro that only gets called for an answered call. I only see this: Executing [91112223...@empl:8] Dial(SIP/mine-0521, Dahdi/G1/111222|60|gM(out-dial)) in new stack DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 -- Requested transfer capability: 0x00 - SPEECH -- Called G1/111222 DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/23 span 1 -- DAHDI/23-1 is proceeding passing it to SIP/mine-0521 -- DAHDI/23-1 is ringing DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on -- DAHDI/23-1 answered SIP/mine-0521 -- Executing [...@macro-out-dial:1] DumpChan(DAHDI/23-1, ) in new stack Dumping Info For Channel: DAHDI/23-1: Info: Name= DAHDI/23-1 Type= DAHDI UniqueID= sys.domain.com-1294514614.2630 CallerID= 9111222 CallerIDName= (N/A) DNIDDigits= (N/A) RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x4 (ulaw) WriteFormat=0x4 (ulaw) ReadFormat= 0x4 (ulaw) 1stFileDescriptor= 35 Framesin= 189 Framesout= 176 TimetoHangup= 0 ElapsedTime=0h0m4s Context=macro-out-dial Extension= s Priority= 1 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: MACRO_DEPTH=1 MACRO_PRIORITY=1 MACRO_CONTEXT=from-outside MACRO_EXTEN= DIALEDPEERNUMBER=G1/111222 TRANSFERCAPABILITY=SPEECH DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0 DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on DAHDI/23-1 DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on DAHDI/23-1 -- Hungup 'DAHDI/23-1' == Spawn extension (empl, 9111222, 8) exited non-zero on 'SIP/mine-0521' -- Executing [...@empl:1] Verbose(SIP/mine-0521, 2|Hangup SIP/mine-0521 with cause 16) in new stack == Hangup SIP/mine-0521 with cause 16 -- Executing [...@empl:2] DumpChan(SIP/mine-0521, ) in new stack Dumping Info For Channel: SIP/mine-0521: Info: Name= SIP/mine-0521 Type= SIP UniqueID= sys.domain.com-1294514614.2629 CallerID= 444555 CallerIDName= Jim Dickenson DNIDDigits= 9111222 RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x2 (gsm) WriteFormat=0x2 (gsm) ReadFormat= 0x2 (gsm) 1stFileDescriptor= 65 Framesin= 248 Framesout= 253 TimetoHangup= 0 ElapsedTime=0h0m0s Context=empl Extension= h Priority= 2 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: DIALSTATUS=ANSWER DIALEDTIME=5 ANSWEREDTIME=1 RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00 BRIDGEPEER=DAHDI/23-1 DIALEDPEERNUMBER=G1/111222 DIALEDPEERNAME=DAHDI/23-1 MACRO_DEPTH=0 RCStatus=0 MyChan=SIP sipcallid=0b69233cd5469...@192.168.0.16 SIPUSERAGENT=Grandstream GXP2000 1.2.2.6 SIPDOMAIN=sys.domain.com SIPURI=sip:m...@00.00.000.000:5064 -- Executing [...@empl:3] ExecIf(SIP/mine-0521, 0|Set|DB(conf//haveadmin)=no) in new stack -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 7, 2011, at 12:44 PM, C F wrote: PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine?
Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
Jim Dickenson wrote:
I am running version 1.4.x. Where do I get PRICAUSE?
NoOP(Hangup Cause: ${HANGUPCAUSE})
Doug
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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine? If a call is placed to a number that is disconnected so a special information tone is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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 -- _ -- 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] Benefit of PRI vs SIP trunk calls
On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote: On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? [snip] I run the PBX for my organization which has about 160 extensions. I wouldn't even think of doing anything but PRI for the main lines because (A) for our size organization where we are located, we're talking a couple hundred dollars a month difference between PRI and SIP in cost so it's nearly break-even in cost which means cost difference isn't a huge motivator, (B) it supports FAX, modems, and TTYs - perfectly, (C) Quality is 100% consistent. In addition, the reliability is good enough that I'm willing to use it for 911. [snip] I have to agree with most of what Joel said in his message. For me, the main problem with many sip implementations is that your phone service will be only as reliable as your internet service. If you have a dedicated internet line that is highly reliable, that's not a big deal, but DSL, Cable, and the like aren't reliable enough for our needs. Having said that, one downside of a PRI is that you are paying for all of those channels, even when you aren't using them. Companies like Paetec and most other large telcos are offering SIP trunks over an MPLS circuit, running on a T1 loop. This covers the reliability problem, as you are running over the same type of circuit as your PRI, and it allows you to take advantage of unused channels as data bandwidth. This is especially helpful for folks who have a data T1 and a PRI, as they can get higher bandwidth for data when there isn't much voice traffic. Because they use G.729, you can also fit more calls on the same circuit. That choice of codec eliminates the ability to send/receive faxes, though, and it's likely expensive when compared to other SIP solutions, but it does appear to be pretty slick. Another benefit of SIP is that it doesn't require a Digium, Sangoma, or similar interface card in the server, simplifying migrations and reducing cost in many scenarios. Tom -- _ -- 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] Benefit of PRI vs SIP trunk calls
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? Assuming you are talking to connect a PBX to the PSTN... PRI advantages: 1. Relatively little equipment between the PTSN and the PBX. Less to break or go wrong. 2. Simple to set up. No need for QoS, routing, authentication, etc. Of course if you only know IP, SIP is easier, but if you learn both, ISDN is easier. 3. If compared to SIP over internet, PRI has guaranteed quality. Granted, SIP *can* have just as good (and better) quality, just not guaranteed if done over the internet (it can be guaranteed over a private circuit). 4. Less latency/delay so there is less talk-over. 5. FAX, high speed modem, TTY, etc, pass-through actually works. (it *can* work over SIP, but Asterisk just isn't quite there yet) I run the PBX for my organization which has about 160 extensions. I wouldn't even think of doing anything but PRI for the main lines because (A) for our size organization where we are located, we're talking a couple hundred dollars a month difference between PRI and SIP in cost so it's nearly break-even in cost which means cost difference isn't a huge motivator, (B) it supports FAX, modems, and TTYs - perfectly, (C) Quality is 100% consistent. In addition, the reliability is good enough that I'm willing to use it for 911. Of course if this installation wasn't in downtown Denver, where ISDN PRI is very cheap (a full CLEC 23-channel ISDN PRI costs roughly what 6 or 7 ILEC POTS lines cost), then SIP would be interested. SIP advantages: 1. Cheap (at least SIP-over-internet) 2. Easy and quick to scale if you have bandwidth. 3. Great for disaster recovery if using SIP over internet 4. Very cheap to get local numbers from all around the world. 5. If using SIP over internet, easy to compare providers -- _ -- 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
