VOIP Tech Chat forum - dslreports.com community

[General] Google+ HangOuts vs. XMPP

2013-05-21 13:46:22

Has anyone here used the Google+ HangOuts? It is supposed to replace GoogleTalk and it just dropped supports for XMPP. Does this mean, GV will soon drop XMPP, too? If so, what will happen to OBi-xx0 devices? -- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together.

[Anveo Direct] Launches Innovative Call Termination Service

2013-05-01 10:47:49

Not often you see a new product announcement which will have a great impact on the future technology and services offered in VOIP market place. Today is one of those days. Back in 2010 Anveo team has launched Anveo.com Retail service which offers unmatched set of features, quality of service, solid infrastructure design and great technology. All of that allows Anveo.com to maintain a technological advantage and ongoing innovations. It is possible due to 'thinking outside the box' approach and highly experienced R&D team. And now with the same spirit Anveo team is launching a new Anveo Direct call termination service which will shake the market with innovative features and flexibility/quality it offers. So what makes Anveo Direct Call Termination service different from the rest of providers? It is our amazing and highly configurable LCR (Least Cost Routing) technology which gives you complete control over how your calls are routed with instant access to over 30 Tier 1 carriers/routes (like Verizon, Comcast, Level 3, TATA Communications, ISP Telecom, BICS, IDT and more). Anveo Direct users have an option of using pre-configured routing options (such as 'lowest cost across all carriers', 'lowest cost across standard routes', 'lowest cost across prime routes') or configure custom routing options and that is where the fun begins :-) Custom routing option allows a user to select which carriers/routes will be used for routing his/her calls, configure automatic failover to the next carrier when one has failed to accept a call, select how many concurrent calls is allowed for a given trunk, select Rate Cap if needed, select the order of carriers when routing calls (by least cost, random order or by service quality). With custom routing options Anveo Direct offers much higher reliability and call delivery success rate than a typical call termination service. Another great feature is 'Route Block'; users can prevent a certain carrier/route from sending calls to a given destination. For example; if end user is reporting DTMF issue on his calls to Rome (Italy), then Anveo Direct user can easily add a Route Block which will prevent future calls to Rome, Italy from being routed though a problematic carrier. Another very handy and unique feature not available from other providers is access to full SIP session trace for any call :-). SIP Session trace is a great tool which helps to investigate interconnect issues. Anveo Direct uses real-time 1 second billing (except for Mexico), does NOT proxy Media (RTP direct from the carrier) and uses IP authentication. Each account can create unlimited number of Call Termination trunks and each Call Termination trunk can have its own dialing prefix, LCR options and the list of authorized IP address. For those interested in some technical details; with over 30 different carriers/routes the number of destinations (price entries for lookups) is so high (close to 3 million entries) that we had to use high performance algorithms and data models to effectively locate routes (based on the phone number dialed). We were able to achieve extremely high lookup speeds and thus VERY low PDD (Post Dial Delay). The average PDD added by Anveo Direct is less than 1/10 of a second!!! For the last 6 month Anveo retail service has been using Anveo Direct for Inbound calls and as of April 2nd Anveo Retail was switched to exclusively use Anveo Direct Outbound Services with outstanding results. I am sure that the launch of Anveo Direct Call Termination services will force other providers to innovate and catch up... We welcome you to try Anveo Direct service with $0.60 test funds at http://www.anveodirect.com

OBi200 now available

2013-05-21 23:41:43

The OBi200 is now available and sells on Amazon for $60. This device has a 1FXS 1USB 1LAN configuration. It supports T.38 and four service providers. said by giqcass :Obi 100 and Obi 202 had a baby? Congratulations! More information: http://www.obitalk.com/forum/index.php?topic=5959.msg38032#msg38032

Will an Asus RT-N66U router work okay with VoIP?

2013-05-04 23:50:12

For various reasons we are thinking about replacing our current router with an Asus RT-N66U. We have a small Asterisk server plus a few VoIP devices (Obihai and Linksys) on the local network and also an Xbox. If anyone is currently using this router, please let me know if you have had any issues with it, particularly related to using it with VoIP and/or an Xbox. If you have any other recommendations for a consumer grade router please feel free to pass them along, or if you know of a good up to date chart of routers that are compatible with VoIP I'd be interested in a link to that. Thanks!

[Equipment] Obi device failures

2013-04-06 17:47:39

How long are you getting your units to last? So far I have two obi 110 with no dialtone even though the account says it is registered and I can make calls using the obitalk app. Also I have a obi100 that has so much noise on the line that no calls can be made. I am getting worried now as I have bought 20 so far and 3 down in less than a year is a bad track record especially since I have bought almost 200 linksys pap2 over the past 5 years and to date only 6 have failed. What gives?

[Anveo] OBi 202+ Anveo - how to disable call waiting?

2013-05-23 19:12:57

Is there a way to disable call waiting for OBi 202 + Anveo combo on per-call basis? Didn't find any relevant * codes on Anveo site and not sure if this is supposed to be Anveo or OBi function.

Bonded DSL, DDR2200, and VOIP

2013-05-23 06:43:13

Is anyone successfully using this combination? I found one topic from a year ago with a similar problem in the CenturyLink forum, but no solution reported and the OP hasn't been back since then. We tried setting up some VOIP phones today over a bonded DSL connection with a Cisco DDR2200 modem/router. This is a Prism IPTV setup, without the TV service activated. The internet works and the modem stats are good on both pairs. It's a 20/2 connection. I get around 19.5 down and 1.5 up on the various speed tests. The DNS resolves and the phones download their config files and images, but are unable to register with the VOIP server. I found the password and was able to login as admin. We disabled the SIP ALG, and tried setting up port forwarding with no luck. The only suggestion CL Support offered was to put the modem in bridge mode and try a separate router. We didn't have another router at the site today, so that's the only thing we didn't try. Any comments or suggestions?

[Asterisk] Freepbx/Asterisk connection issue

2013-05-23 09:41:42

Good morning, I'm using Freepbx/Asterisk flavor for my voip at home. It work fine (with +-10 voip cisco phones) Sometime, my internet connection drop, and reconnect itself (using FTTH connection (pppoe), with a Juniper SSG140 as my router. When my connection drop, by pbx does not reconnect itself to my provider (voip.ms). I have to manually reboot my box to have allow the pbx to reconnect.. any thing that can be done to have it reconnect itself without doing that? Thanks Frank -- ---- On Rogers with a Blackberry z10

(Voip.ms) Can you receive CallerID blocked calls?

2013-05-23 11:13:10

I've been having problems with my Voip.ms NJ DID since Monday. I've changed servers and Points of Presence, because callers were complaining about getting a busy signal or going straight to voice-mail. After a couple of days with everything working properly, people are complaining about getting a busy signal. I called my DID using my home & cell phones and got the busy signal, too. I block my outgoing CallerID (yes - I know!), and unblocked and tried again. I got through this time. The strange thing is that I've never had a problem calling my Voip.ms DID using my home or cell phones before (with outgoing CallerID blocked). Can anyone else try blocking their outgoing number and calling their DID? Does it ring through or give a busy signal? Thanks, Mike

SPA 2102 locked on comwave

2013-04-25 21:14:47

Hello, I have recently found in recycling ( Earth day was here in Montreal) a SPA 2102 brand new ( in the original plastic bag). I was happy to try to configure until I found that is requesting admin pass. I tried some of the google resources, did a wireshark packet inspection while it was connected to a bridged connection in a laptop, so it will get an IP through the wifi, it tries to resolve the dmsc.comwave.net and then connects to 209.47.87.38, couple of handshakes, it gets kicked out, but nothing more, than again same thing. There is no explicit file or whatsoever config the box tries to get. I’m stuck after 3 days of unsuccessfully trials to configure a dns and dhcp to see if it gets further with the connection. Does someone knows more on how should I start a setup to make a DNS working and then a DHCP, ( on XP or 7), do I need a router?, or a switch and 2 computers?, I’m lost here. Or someone knows the password for recent comwave locked configs or some reset code? I have seen Toro still interested in these things, a while back I had unlocked my Vonage box with the serial cable, too bad that after 1.5 years it died last year (flash chip went bad, rebooting). Since then I went to unlocked SPA2102, works fine until now, but I was hoping to make a spare after the last experience with VDV21. Thanks

[Equipment] OBi110 built-in Voicemail?

2013-03-24 08:33:56

The other day, I called my friend's GV (configured on an OBi-110 device) and was greeted by a maie voicemail IVR. I didn't leave a message, but wonder if that IVR is a build-in feature on the OBi-110 device. AFAIK, my friend has not activated the OBi-110 device to use OBiTalk services. Can any OBi-110 owners here confirm that an OBi-110 has a built-in voicemail feature? -- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together.

[Equipment] call cuts with a maximum of 1minute to 2minutes

2013-05-22 12:02:12

Hy.. anyone assist. I am a remote support tech. supporting a Gigaset A510IP for my client. Well.. they have a new splitter, Telephony set correct on BASE, Audio(checked Audio and RFC2388). port forwarding on router is set 5060 enable. domain is set correct, RTP on my billing system is OK. now Problem>>> calls cut Telkom(ADSL provider) tested line was faulty, they fixed it, raised line speed from 1Mbps to 4Mbps.. what the last important setting am i missing here? :huh:

Weekly VoIP Discussions

2013-05-23 01:52:03

For the past several months, VoIP Users Conference (weekly since 2007) has been testing the Google Plus and Youtube live video streaming. During these months, Google Plus has changed many things and so has Youtube. I'm hoping most of the things are now settled enough for me to invite you to join us on these. If you are interested, you are welcome to subscribe to the Youtube channel 'voipusers' and/or join the G+ community, now called IP Communications & VoIP, which covers all the exciting things coming out there days, such as WebRTC, HD Voice and new codecs, etc. AUDIO - The way the calls work now is that g722 audio on a SIP conference server is bridged with the Google Hangout creating a pretty cool event. You can call in via SIP or PSTN (Skype even). VIDEO is visible live or recorded on Youtube. You can get in the Hangout by asking for an invitation on G+ in the community (up to 9 people). After the first hour which is usually devoted to a guest, the conference goes on in audio for as long as people are talking. QUESTIONS are welcome, this is a real community, many of us have met in person at shows like Astricon, and the collective knowledge and experience of participants are amazing. As a long time member of this forum, I really hope some of the regulars will join us Fridays. Beginners are welcome, too. BETA programs are often available for testing new SIP or video clients, mobile apps, etc. We work with the industry, giving them valuable feedback. YOU can be a part of this. -- VoIP Users Conference http://vuc.me

Cream pie and VoIP

2013-05-23 00:07:07

This is the DSLR Voip Tech Chat forum. For a few years there was a totally different site called [Voip Tech Chat], run by two guys named Patrick and Fred. (Fred Posner is a VoIP professional).... ....and the website was oddly enough located at VoipTechChat.com Fred and Patrick haven't been doing that lately. Now here is my question: If you go to http://voiptechchat.com/ Previously that went to their VoIP website, but NOW it goes to a BAKERY. Anybody with any ideas as to how this happens?

FoIP server

2013-05-15 17:32:56

http://www.dslreports.com/forum/r28289721-Foip-server anyone have experience with setting up a FoIP server and which one would be best? I'm looking into replacing a multitech faxfinder (that uses 4 analog lines) with a VM that would communicate with a Cisco VoIP system I'm looking for it to do e-mail to fax and fax to e-mail the current device uses a windows app to send and faxes it receives are sent to e-mail (we have 100 DID's for fax use) (so one thats relatively easy to bulk setup would be nice.) -- http://www.change.org/petitions/create-a-100-offline-single-player-mode-in-simcity-2013-remove-the-origin-requirement-from-it-and-bring-back-popular-features-from-simcity-4

Caller ID Disa/Forwarding never works with Asterisk/voip.ms

2013-05-20 10:58:14

I've posted this before but never got it resolved. Trying to set up a test server but for some reason I can never get (Canadian style) caller ID to work properly with Disa, forwarding etc. My setup is: 1) Hosted Asterisk PBX running PIAF Purple 2) Connected to a voip.ms trunk for inbound/outbound with routes assigned 3) DIDs with voip.ms pointing to my Asterisk System 4) A DISA Set up as a test that has Caller ID set as "Sunshine Tech" , as per the help bubble it says "User Name" 33333 etc. No matter what I try, I can never get it to display the caller ID on the outbound DISA and can't figure out why. Below is the SIP log of me calling into a voip.ms DID 647-478-1010 which is set to go right to DISA, from my cell phone 416-122-1199 and then dialing 18662223456 as a test. No matter what, the voip.ms log, and the person I'm calling always just sees the NAME as 6475576644 and Number as 6475576644 it never shows "Sunshine Tech" in this example as the name. This is the same with if I even just set an extension to forward to my cell, it never keeps the persons name, it only shows their phone # as the name and number. I can't figure out for the life of me why it won't send the name but voip.ms says it's a problem with my config. However I have installed PIAF, FreePBX, AsteriskNow, literally every distribution and it always does it, so either it has to be just not supported or it has to be some kind of setting? .. -- Executing [6474781010@from-trunk:1] Set("SIP/voipms-00000021", "__FROM_DID=6474781010") in new stack -- Executing [6474781010@from-trunk:2] Gosub("SIP/voipms-00000021", "app-blacklist-check,s,1()") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("SIP/voipms-00000021", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("SIP/voipms-00000021", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("SIP/voipms-00000021", "") in new stack -- Executing [6474781010@from-trunk:3] Set("SIP/voipms-00000021", "CDR(did)=6474781010") in new stack -- Executing [6474781010@from-trunk:4] ExecIf("SIP/voipms-00000021", "0 ?Set(CALLERID(name)=4161221199)") in new stack -- Executing [6474781010@from-trunk:5] Set("SIP/voipms-00000021", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [6474781010@from-trunk:6] Set("SIP/voipms-00000021", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [6474781010@from-trunk:7] Goto("SIP/voipms-00000021", "disa,1,1") in new stack -- Goto (disa,1,1) -- Executing [1@disa:1] Authenticate("SIP/voipms-00000021", "336511,") in new stack -- Playing 'agent-pass.gsm' (language 'en') -- Playing 'auth-thankyou.gsm' (language 'en') -- Executing [1@disa:2] Set("SIP/voipms-00000021", "_DISA=disa^1^newcall") in new stack -- Executing [1@disa:3] Set("SIP/voipms-00000021", "_DISACONTEXT=from-internal") in new stack -- Executing [1@disa:4] Set("SIP/voipms-00000021", "_KEEPCID=TRUE") in new stack -- Executing [1@disa:5] Set("SIP/voipms-00000021", "_HANGUP=") in new stack -- Executing [1@disa:6] Set("SIP/voipms-00000021", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5.000 -- Executing [1@disa:7] Set("SIP/voipms-00000021", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10.000 -- Executing [1@disa:8] Set("SIP/voipms-00000021", "CALLERID(all)="Sunshine Tech" <6475576644>") in new stack -- Executing [1@disa:9] Set("SIP/voipms-00000021", "__REALCALLERIDNUM=6475576644") in new stack -- Executing [1@disa:10] DISA("SIP/voipms-00000021", "no-password,disa-dial") in new stack -- Executing [18662223456@disa-dial:1] NoOp("SIP/voipms-00000021", "called 18662223456 in from-internal by ID: 1") in new stack -- Executing [18662223456@disa-dial:2] Dial("SIP/voipms-00000021", "Local/18662223456@from-internal,300,") in new stack -- Called Local/18662223456@from-internal -- Executing [18662223456@from-internal:1] Macro("Local/18662223456@from-internal-00000006;2", "user-callerid,LIMIT,") in new stack -- Executing [s@macro-user-callerid:1] Set("Local/18662223456@from-internal-00000006;2", "AMPUSER=6475576644") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("Local/18662223456@from-internal-00000006;2", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(REALCALLERIDNUM=6475576644)") in new stack -- Executing [s@macro-user-callerid:4] Set("Local/18662223456@from-internal-00000006;2", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("Local/18662223456@from-internal-00000006;2", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("Local/18662223456@from-internal-00000006;2", "1?report") in new stack -- Goto (macro-user-callerid,s,13) -- Executing [s@macro-user-callerid:13] GotoIf("Local/18662223456@from-internal-00000006;2", "1?continue") in new stack -- Goto (macro-user-callerid,s,26) -- Executing [s@macro-user-callerid:26] Set("Local/18662223456@from-internal-00000006;2", "CALLERID(number)=6475576644") in new stack -- Executing [s@macro-user-callerid:27] Set("Local/18662223456@from-internal-00000006;2", "CALLERID(name)=Sunshine Tech") in new stack -- Executing [s@macro-user-callerid:28] Set("Local/18662223456@from-internal-00000006;2", "CHANNEL(language)=en") in new stack -- Executing [18662223456@from-internal:2] Set("Local/18662223456@from-internal-00000006;2", "MOHCLASS=default") in new stack -- Executing [18662223456@from-internal:3] Set("Local/18662223456@from-internal-00000006;2", "_NODEST=") in new stack -- Executing [18662223456@from-internal:4] Gosub("Local/18662223456@from-internal-00000006;2", "sub-record-check,s,1(out,18662223456,)") in new stack -- Executing [s@sub-record-check:1] GotoIf("Local/18662223456@from-internal-00000006;2", "1?check") in new stack -- Goto (sub-record-check,s,6) -- Executing [s@sub-record-check:6] Set("Local/18662223456@from-internal-00000006;2", "__MON_FMT=wav") in new stack -- Executing [s@sub-record-check:7] GotoIf("Local/18662223456@from-internal-00000006;2", "1?next") in new stack -- Goto (sub-record-check,s,10) -- Executing [s@sub-record-check:10] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Return()") in new stack -- Executing [s@sub-record-check:11] GotoIf("Local/18662223456@from-internal-00000006;2", "0?out,1") in new stack -- Executing [s@sub-record-check:12] Set("Local/18662223456@from-internal-00000006;2", "__REC_STATUS=INITIALIZED") in new stack -- Executing [s@sub-record-check:13] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(__REC_POLICY_MODE=)") in new stack -- Executing [s@sub-record-check:14] Set("Local/18662223456@from-internal-00000006;2", "NOW=1369061565") in new stack -- Executing [s@sub-record-check:15] Set("Local/18662223456@from-internal-00000006;2", "__DAY=20") in new stack -- Executing [s@sub-record-check:16] Set("Local/18662223456@from-internal-00000006;2", "__MONTH=05") in new stack -- Executing [s@sub-record-check:17] Set("Local/18662223456@from-internal-00000006;2", "__YEAR=2013") in new stack -- Executing [s@sub-record-check:18] Set("Local/18662223456@from-internal-00000006;2", "__TIMESTR=20130520-075245") in new stack -- Executing [s@sub-record-check:19] Set("Local/18662223456@from-internal-00000006;2", "__FROMEXTEN=6475576644") in new stack -- Executing [s@sub-record-check:20] Set("Local/18662223456@from-internal-00000006;2", "__CALLFILENAME=out-18662223456-6475576644-20130520-075245-1369061565.48") in new stack -- Executing [s@sub-record-check:21] Goto("Local/18662223456@from-internal-00000006;2", "out,1") in new stack -- Goto (sub-record-check,out,1) -- Executing [out@sub-record-check:1] ExecIf("Local/18662223456@from-internal-00000006;2", "1?Set(__REC_POLICY_MODE=)") in new stack -- Executing [out@sub-record-check:2] GosubIf("Local/18662223456@from-internal-00000006;2", "0?record,1(exten,18662223456,6475576644)") in new stack -- Executing [out@sub-record-check:3] Return("Local/18662223456@from-internal-00000006;2", "") in new stack -- Executing [18662223456@from-internal:5] Macro("Local/18662223456@from-internal-00000006;2", "dialout-trunk,1,18662223456,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("Local/18662223456@from-internal-00000006;2", "DIAL_TRUNK=1") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("Local/18662223456@from-internal-00000006;2", "0?sub-pincheck,s,1()") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("Local/18662223456@from-internal-00000006;2", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("Local/18662223456@from-internal-00000006;2", "DIAL_NUMBER=18662223456") in new stack -- Executing [s@macro-dialout-trunk:5] Set("Local/18662223456@from-internal-00000006;2", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("Local/18662223456@from-internal-00000006;2", "OUTBOUND_GROUP=OUT_1") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("Local/18662223456@from-internal-00000006;2", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("Local/18662223456@from-internal-00000006;2", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("Local/18662223456@from-internal-00000006;2", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("Local/18662223456@from-internal-00000006;2", "outbound-callerid,1") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("Local/18662223456@from-internal-00000006;2", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(REALCALLERIDNUM=6475576644)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("Local/18662223456@from-internal-00000006;2", "0?normcid") in new stack -- Executing [s@macro-outbound-callerid:4] Set("Local/18662223456@from-internal-00000006;2", "USEROUTCID=6475576644") in new stack -- Executing [s@macro-outbound-callerid:5] GotoIf("Local/18662223456@from-internal-00000006;2", "1?bypass") in new stack -- Goto (macro-outbound-callerid,s,7) -- Executing [s@macro-outbound-callerid:7] Set("Local/18662223456@from-internal-00000006;2", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("Local/18662223456@from-internal-00000006;2", "TRUNKOUTCID=6474789362") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("Local/18662223456@from-internal-00000006;2", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("Local/18662223456@from-internal-00000006;2", "1?Set(CALLERID(all)=6474789362)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("Local/18662223456@from-internal-00000006;2", "1?Set(CALLERID(all)=6475576644)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("Local/18662223456@from-internal-00000006;2", "0?sub-flp-1,s,1()") in new stack -- Executing [s@macro-dialout-trunk:13] Set("Local/18662223456@from-internal-00000006;2", "OUTNUM=18662223456") in new stack -- Executing [s@macro-dialout-trunk:14] Set("Local/18662223456@from-internal-00000006;2", "custom=SIP/voipms") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack -- Executing [s@macro-dialout-trunk:17] Macro("Local/18662223456@from-internal-00000006;2", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("Local/18662223456@from-internal-00000006;2", "") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("Local/18662223456@from-internal-00000006;2", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:19] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(CONNECTEDLINE(num,i)=18662223456)") in new stack -- Executing [s@macro-dialout-trunk:20] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(CONNECTEDLINE(name,i)=CID:6475576644)") in new stack -- Executing [s@macro-dialout-trunk:21] GotoIf("Local/18662223456@from-internal-00000006;2", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:22] Dial("Local/18662223456@from-internal-00000006;2", "SIP/voipms/18662223456,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/voipms/18662223456 -- SIP/voipms-00000022 is making progress passing it to Local/18662223456@from-internal-00000006;2 -- Local/18662223456@from-internal-00000006;1 is making progress passing it to SIP/voipms-00000021 -- SIP/voipms-00000022 answered Local/18662223456@from-internal-00000006;2 -- Local/18662223456@from-internal-00000006;1 answered SIP/voipms-00000021 -- Executing [h@macro-dialout-trunk:1] Macro("Local/18662223456@from-internal-00000006;2", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/18662223456@from-internal-00000006;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf("Local/18662223456@from-internal-00000006;2", "0?Set(CDR(recordingfile)=)") in new stack -- Executing [s@macro-hangupcall:4] Hangup("Local/18662223456@from-internal-00000006;2", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Local/18662223456@from-internal-00000006;2' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'Local/18662223456@from-internal-00000006;2' == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'Local/18662223456@from-internal-00000006;2' in macro 'dialout-trunk' == Spawn extension (from-internal, 18662223456, 5) exited non-zero on 'Local/18662223456@from-internal-00000006;2' -- Executing [h@disa-dial:1] NoOp("SIP/voipms-00000021", "called h in from-internal by ID: 1") in new stack -- Executing [h@disa-dial:2] Dial("SIP/voipms-00000021", "Local/h@from-internal,300,") in new stack -- Called Local/h@from-internal -- Executing [h@from-internal:1] Hangup("Local/h@from-internal-00000007;2", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/h@from-internal-00000007;2' -- Executing [h@from-internal:1] Hangup("Local/h@from-internal-00000007;2", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/h@from-internal-00000007;2' -- No one is available to answer at this time (1:0/0/0) -- Executing [h@disa-dial:3] Gosub("SIP/voipms-00000021", "s-NOANSWER,1()") in new stack -- Executing [s-NOANSWER@disa-dial:1] NoOp("SIP/voipms-00000021", "DISA Dial failed due to NOANSWER - returning to dial tone") in new stack -- Executing [s-NOANSWER@disa-dial:2] PlayTones("SIP/voipms-00000021", "congestion") in new stack -- Executing [s-NOANSWER@disa-dial:3] Wait("SIP/voipms-00000021", "3") in new stack == Spawn extension (disa-dial, s-NOANSWER, 3) exited non-zero on 'SIP/voipms-00000021' == Spawn extension (disa-dial, 18662223456, 2) exited non-zero on 'SIP/voipms-00000021'

[CallCentric] Perhaps hidden setting affecting nonregistration f

2013-05-21 16:50:37

EDIT: Truncated title is supposed to read: " Perhaps hidden setting affecting nonregistration failover " =================================== I stumbled across something that I have not seen mention of (although I could have missed it). I was tweaking my CC Call Treatments today. I've always had one that redirects calls to my cellphone if the registration to the ATA serving my home primary line fails. I got to thinking: now that "extensions" have been implemented, how does that impact my non-registration redirection? So, I looked at that call treatment, and sure enough there are new options dealing with extensions, and they defaulted to something that I did not want and needed to fix. After "extensions" were implemented I had quickly set-up 3 new ones in addition to my original home ATA: one for my iPod SIP client; one for my PBXes account; and one for a laptop softphone. In my "not-registered; fail-over to cellphone number" Call Treatment, things looked like this: Status is: - Any - I'm on a call X Not registered on the following extensions: X All my extensions - 100 - Home - 101 - iPod - 102 - PBXes - 103 - Laptop Not what I want! That logic won't fail-over to my cell unless ALL extensions are unregistered. Unlikely; the PBXes extension won't easily unregister, but it doesn't ring any convenient phone. What I =want= so as to replicate what I had before is to uncheck "all my extensions" and check only "home." Then, if there is a failure of registration (due to power failure; internet failure; whatever) of my home CallCentric phone, my cell will ring. Maybe obvious to some, but it wasn't to me...

Rebtel &raquo; Skype -> SIP

2013-04-13 13:03:53

I have been utilizing Skype to SIP setup per http://www.dslreports.com/forum/r24541269- for two years without changing anything. Recently, Skype callers told me that when they called me at rebtel_skype_buddy01, they got "If you are member press 1, for help press 9", and my phone never ring. Can others confirm this setup no longer working? If this is the case, what will be the best and simplest alternative?

pbxinaflash.com/community site down?

2013-05-16 13:50:03

Error message [404] 404 Not Found for pbxinaflash.com/community/index.php port 80 on Thursday, 16-May-2013 12:48:10 CDT Anyone else seeing this? -- (1) It's either 99¢ or $0.99; not .99¢ (2) It's "so MUCH fun" not "so fun"

[Voip.ms] Is there a minute cap on sip calls? 9:1 free vs paid

2013-05-20 21:50:36

I share my account with 2 others and between us there's nearly 54 hours of sip calls (we use IPKall alot) compared to just under 6 hours of pstn calls in the same time period. I was wondering, does anyone know if there's a limit or at what point does it become excessive? I know every call uses bandwidth and other resources, but when does it become too much? -- ...Who, What, When, Where, How... Why? Why Not?