- VoiceOps - lists.voiceops.org

TA900 DNS problems
by mylists＠battleop.com Sept. 18, 2011

Sept. 18, 2011

We changed the IPs on our sip servers last night. A few days before we changed them we made the following change in our TA900s: voice trunk T01 type sip sip-server primary 216.x.x.x registrar primary 216.x.x.x to voice trunk T01 type sip sip-server primary sip3.atlga.domain.com registrar primary sip3.atlga.domain.com Then we changed the A record from the old IP to the new IP. We run a pretty low TTL on that domain and flushed the DNS servers that the TA900s look to. After several hours about 60% of the TA900s required a reboot to resolve the hostname to the new IP and the other 40% won't do it no matter how many times your reboot. The only fix is to change everything back to IP based using the new IPs. The oldest firmware we are running is T900G2A-A4-02-00-E.biz upto the most recent. Has anyone else run into this kind of trouble with the TA900s? Richey

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Calls from 118772988855
by matthew＠corp.crocker.com Sept. 16, 2011

Sept. 16, 2011

I'm seeing a lot of calls from 118772988855 hitting my customers. I actually had two hit my home line today simultaneously. It was a 'you could get a better credit card interest rate' scam. The calls came in through my Verizon SS7 interconnect. Does anyone know the source carrier? Any way to track them down. Thanks -Matt -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760

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Fax Server Solutions
by RDawson＠alliedtelecom.net Sept. 16, 2011

Sept. 16, 2011

Anyone using FaxBack Satisfaxtion? Any pros/cons/thoughts? Particularly interested in feedback on the HTTPS ATA solution. Any other alternative suggestions? [Description: cid:image001.jpg at 01CBD82E.96AB3910] Robert Dawson Voice Systems Engineer 202-349-0477(o) | 571-234-2621 (c) | 202-541-9050 (f) rdawson at alliedtelecom.net<mailto:rdawson at alliedtelecom.net>~ www.alliedtelecom.com 1120 20th St NW Suite 500-S Washington, DC 20036

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Asterisk 1.8.6.0 unable to allocate a port for RTP instance
by recourse＠gmail.com Sept. 16, 2011

Sept. 16, 2011

Hi list! On the first we upgraded one of our Asterisk boxes from 1.4 to 1.8.6.0 everything ran OK until tonight we started getting the following messages on the asterisk console and in the log file. [Sep 15 18:05:21] ERROR[14306] res_rtp_asterisk.c: Oh dear... we couldn't allocate a port for RTP instance '0xb138$ [Sep 15 18:05:21] ERROR[14306] chan_sip.c: Got SDP but have no RTP session allocated. Looked at the active number of channels using core show channels and no channels were reported as being in use/active. had to restart the asterisk service to restore access. No other messages in the log to indicate what could be the issue. Anyone else ever see that happen? Thanks! Joseph

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help
by apaminerala＠yahoo.com Sept. 15, 2011

Sept. 15, 2011

? Tudor Seicarescu Saving money for business worldwide Tel. Montreal: 1-514-400-2575 Tel. Romania: 40-21-589-7246 Cell Romania: 40-762-863-429 Yahoo: apaminerala Skype: apaminerala http://ca.linkedin.com/in/apaminerala ________________________________ From: "voiceops-request at voiceops.org" <voiceops-request at voiceops.org> To: voiceops at voiceops.org Sent: Wednesday, September 14, 2011 8:22 AM Subject: VoiceOps Digest, Vol 27, Issue 10 Send VoiceOps mailing list submissions to ??? voiceops at voiceops.org To subscribe or unsubscribe via the World Wide Web, visit ??? https://puck.nether.net/mailman/listinfo/voiceops or, via email, send a message with subject or body 'help' to ??? voiceops-request at voiceops.org You can reach the person managing the list at ??? voiceops-owner at voiceops.org When replying, please edit your Subject line so it is more specific than "Re: Contents of VoiceOps digest..." Today's Topics: ? 1. Re: Cell Phone to SIP provider (Matthew S. Crocker) ? 2. Re: Cell Phone to SIP provider (Carlos Alvarez) ---------------------------------------------------------------------- Message: 1 Date: Wed, 14 Sep 2011 07:24:32 -0400 (EDT) From: "Matthew S. Crocker" <matthew at corp.crocker.com> To: Paul Timmins <paul at timmins.net> Cc: "VoiceOps at voiceops.org" <voiceops at voiceops.org> Subject: Re: [VoiceOps] Cell Phone to SIP provider Message-ID: <a2e95fd1-297b-44d5-a9d3-9f6ad2644f34 at zimbra1.crocker.com> Content-Type: text/plain; charset="utf-8" Of course there is no reason why SIP couldn't be used for signaling and just establish the voice over a GSM / CDMA codec. The cell tower could transcode to RTP. Getting SIP direct to the handset would be good . The cell towers would just need to act like a SIP proxy ----- Original Message ----- > From: "Paul Timmins" <paul at timmins.net> > To: "Scott Berkman" <scott at sberkman.net> > Cc: "VoiceOps at voiceops.org" <voiceops at voiceops.org> > Sent: Wednesday, September 14, 2011 12:07:14 AM > Subject: Re: [VoiceOps] Cell Phone to SIP provider > My android phones do SIP user interface integration over HSDPA and > EVDO respectively, and frankly, the voice quality isn't that great. > Much worse than the underlying GSM or CDMA signal bar for signal > bar. > There's a reason the carriers aren't doing this yet. > -Paul > On Sep 13, 2011, at 11:32 PM, Scott Berkman wrote: > > You might want to see if LightSquared has anything for you, they > > dub > > themselves the ?only wholesale-only mobile carrier?, but I?m not > > sure what they can offer today. > > > http://www.lightsquared.com/ > > > Also, aren?t some of the newer cellular technologies (UMTS or EVDO) > > inherently built on SIP for the voice portion of the network? > > That?s > > not an answer to your problem, but? > > > One other thing you might want to look into is an OS that allows > > the > > default phone app to use SIP over the data connection natively (as > > opposed to using a separate soft phone application that might not > > integrate as well to the UI). I believe this was preset in versions > > of WM6, but that?s somewhat old now and I have no idea if it?s > > built > > into any other platforms. > > > http://voipguides.blogspot.com/2008/05/voip-sip-client-for-windows-mobile-6… > > > -Scott > > > From: voiceops-bounces at voiceops.org > > [mailto:voiceops-bounces at voiceops.org] On Behalf Of Carlos Alvarez > > > Sent: Tuesday, September 13, 2011 3:44 PM > > > To: VoiceOps at voiceops.org > > > Subject: Re: [VoiceOps] Cell Phone to SIP provider > > > It is my understanding that: > > > a. The original post asked about sending the calls to the end point > > (cell phone) as SIP, not cellular voice. > > > b. Sprint's MIS (Mobile Integration Service) merely allows a > > company > > or service provider to connect to them directly over TDM or SIP and > > do seamless integration of the cell and other phones, but the call > > is still cellular voice to the handset. > > > I am facing a competitive sale right now to a municipality where > > there is only one other carrier who can compete with us, and only > > on > > that one feature. > > > On Tue, Sep 13, 2011 at 12:33 PM, Matthew S. Crocker < > > matthew at corp.crocker.com > wrote: > > > Sprint does this. The cell phones register to the cell tower as > > cell > > phones. The cell tower converts to SIP and registers with the call > > agent. Broadsoft mentioned something about this at the last > > connections conference. You would get a private IP connection into > > Sprint to carry the voice traffic. This would be perfect for me if > > Sprint had a decent cell network in my area. Does anyone know if > > AT&T or Verizon (LTE) do it as well? > > > http://newsroom.sprint.com/article_display.cfm?article_id=1628 > > > > From: "Oren Yehezkely" < orenyny at gmail.com > > > > > > > To: " VoiceOps at voiceops.org " < voiceops at voiceops.org > > > > > > > Sent: Tuesday, September 13, 2011 1:00:02 PM > > > > > > Subject: [VoiceOps] Cell Phone to SIP provider > > > > > > Hello, > > > > > > I am looking for a cell phone company that is capable of > > > providing > > > SIP service instead of sending the calls over the cellular > > > network. > > > > > > If anybody here is capable of providing such a service or can > > > recommend a vendor that can do that, please let me know. > > > > > > Thanks, > > > > > > Oren > > > > > > _______________________________________________ > > > > > > VoiceOps mailing list > > > > > > VoiceOps at voiceops.org > > > > > > https://puck.nether.net/mailman/listinfo/voiceops > > > > > _______________________________________________ > > > VoiceOps mailing list > > > VoiceOps at voiceops.org > > > https://puck.nether.net/mailman/listinfo/voiceops > > > -- > > > Carlos Alvarez > > > TelEvolve > > > 602-889-3003 > > > _______________________________________________ > > > VoiceOps mailing list > > > VoiceOps at voiceops.org > > > https://puck.nether.net/mailman/listinfo/voiceops > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops

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help
by apaminerala＠yahoo.com Sept. 15, 2011

Sept. 15, 2011

? Tudor Seicarescu Saving money for business worldwide Tel. Montreal: 1-514-400-2575 Tel. Romania: 40-21-589-7246 Cell Romania: 40-762-863-429 Yahoo: apaminerala Skype: apaminerala http://ca.linkedin.com/in/apaminerala ________________________________ From: "voiceops-request at voiceops.org" <voiceops-request at voiceops.org> To: voiceops at voiceops.org Sent: Wednesday, September 14, 2011 9:57 AM Subject: VoiceOps Digest, Vol 27, Issue 11 Send VoiceOps mailing list submissions to ??? voiceops at voiceops.org To subscribe or unsubscribe via the World Wide Web, visit ??? https://puck.nether.net/mailman/listinfo/voiceops or, via email, send a message with subject or body 'help' to ??? voiceops-request at voiceops.org You can reach the person managing the list at ??? voiceops-owner at voiceops.org When replying, please edit your Subject line so it is more specific than "Re: Contents of VoiceOps digest..." Today's Topics: ? 1. Re: Cell Phone to SIP provider (Paul Timmins) ---------------------------------------------------------------------- Message: 1 Date: Wed, 14 Sep 2011 12:56:47 -0400 From: Paul Timmins <paul at timmins.net> To: voiceops at voiceops.org Subject: Re: [VoiceOps] Cell Phone to SIP provider Message-ID: <4E70DCCF.7070209 at timmins.net> Content-Type: text/plain; charset="windows-1252"; Format="flowed" That's like saying that ISDN lines and SIP are the same thing because they both use G.711 ulaw with common channel signalling, and it's really up to the phone companies they could be using SIP over ISDN instead. The difference is that the digital data you speak of is using codecs designed for high loss and corruption, has a delay certain model (unlike packet switched data) and doesn't have overhead (like IP). -Paul On 09/14/2011 11:22 AM, Carlos Alvarez wrote: > You seem to be suggesting that there is a technical reason the > carriers don't do this.? There is not.? Carriers prioritize voice > channels, which really are just data also, though not using SIP.? They > *could* choose to prioritize data, or do the usual QoS for the data so > VoIP has good quality, just like any other IP connection.? They won't, > because it's a huge threat to their business model. > > > On Tue, Sep 13, 2011 at 9:07 PM, Paul Timmins <paul at timmins.net > <mailto:paul at timmins.net>> wrote: > >? ? My android phones do SIP user interface integration over HSDPA and >? ? EVDO respectively, and frankly, the voice quality isn't that >? ? great. Much worse than the underlying GSM or CDMA signal bar for >? ? signal bar. > >? ? There's a reason the carriers aren't doing this yet. >? ? -Paul > >? ? On Sep 13, 2011, at 11:32 PM, Scott Berkman wrote: > >>? ? You might want to see if LightSquared has anything for you, they >>? ? dub themselves the ?only wholesale-only mobile carrier?, but I?m >>? ? not sure what they can offer today. >>? ? http://www.lightsquared.com/ >>? ? Also, aren?t some of the newer cellular technologies (UMTS or >>? ? EVDO) inherently built on SIP for the voice portion of the >>? ? network?? That?s not an answer to your problem, but? >>? ? One other thing you might want to look into is an OS that allows >>? ? the default phone app to use SIP over the data connection >>? ? natively (as opposed to using a separate soft phone application >>? ? that might not integrate as well to the UI).? I believe this was >>? ? preset in versions of WM6, but that?s somewhat old now and I have >>? ? no idea if it?s built into any other platforms. >>? ? http://voipguides.blogspot.com/2008/05/voip-sip-client-for-windows-mobile-6… >>? ? -Scott >>? ? *From:*voiceops-bounces at voiceops.org >>? ? <mailto:voiceops-bounces at voiceops.org> >>? ? [mailto:voiceops-bounces at voiceops.org >>? ? <mailto:voiceops-bounces at voiceops.org>]*On Behalf Of*Carlos Alvarez >>? ? *Sent:*Tuesday, September 13, 2011 3:44 PM >>? ? *To:*VoiceOps at voiceops.org <mailto:VoiceOps at voiceops.org> >>? ? *Subject:*Re: [VoiceOps] Cell Phone to SIP provider >>? ? It is my understanding that: >>? ? a.? The original post asked about sending the calls to the end >>? ? point (cell phone) as SIP, not cellular voice. >>? ? b.? Sprint's MIS (Mobile Integration Service) merely allows a >>? ? company or service provider to connect to them directly over TDM >>? ? or SIP and do seamless integration of the cell and other phones, >>? ? but the call is still cellular voice to the handset. >>? ? I am facing a competitive sale right now to a municipality where >>? ? there is only one other carrier who can compete with us, and only >>? ? on that one feature. >> >>? ? On Tue, Sep 13, 2011 at 12:33 PM, Matthew S. Crocker >>? ? <matthew at corp.crocker.com <mailto:matthew at corp.crocker.com>> wrote: >> >>? ? Sprint does this.? The cell phones register to the cell tower as >>? ? cell phones.? The cell tower converts to SIP and registers with >>? ? the call agent.? Broadsoft mentioned something about this at the >>? ? last connections conference.? You would get a private IP >>? ? connection into Sprint to carry the voice traffic.? ? This would >>? ? be perfect for me if Sprint had a decent cell network in my >>? ? area.? Does anyone know if AT&T or Verizon (LTE) do it as well? >> >>? ? http://newsroom.sprint.com/article_display.cfm?article_id=1628 >> >>? ? ------------------------------------------------------------------------ >> >>? ? ? ? *From:*"Oren Yehezkely" <orenyny at gmail.com >>? ? ? ? <mailto:orenyny at gmail.com>> >>? ? ? ? *To:*"VoiceOps at voiceops.org <mailto:VoiceOps at voiceops.org>" >>? ? ? ? <voiceops at voiceops.org <mailto:voiceops at voiceops.org>> >>? ? ? ? *Sent:*Tuesday, September 13, 2011 1:00:02 PM >>? ? ? ? *Subject:*[VoiceOps] Cell Phone to SIP provider >> >> >> >>? ? ? ? Hello, >> >>? ? ? ? I am looking for a cell phone company that is capable of >>? ? ? ? providing SIP service instead of sending the calls over the >>? ? ? ? cellular network. >> >>? ? ? ? If anybody here is capable of providing such a service or can >>? ? ? ? recommend a vendor that can do that, please let me know. >> >>? ? ? ? Thanks, >> >>? ? ? ? Oren >> >>? ? ? ? _______________________________________________ >>? ? ? ? VoiceOps mailing list >>? ? ? ? VoiceOps at voiceops.org <mailto:VoiceOps at voiceops.org> >>? ? ? ? https://puck.nether.net/mailman/listinfo/voiceops >> >> >>? ? _______________________________________________ >>? ? VoiceOps mailing list >>? ? VoiceOps at voiceops.org <mailto:VoiceOps at voiceops.org> >>? ? https://puck.nether.net/mailman/listinfo/voiceops >> >> >> >>? ? -- >>? ? Carlos Alvarez >>? ? TelEvolve >>? ? 602-889-3003 <tel:602-889-3003> >>? ? _______________________________________________ >>? ? VoiceOps mailing list >>? ? VoiceOps at voiceops.org <mailto:VoiceOps at voiceops.org> >>? ? https://puck.nether.net/mailman/listinfo/voiceops > > > > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops

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Verizon or Level3 in the DC/Alexandria VA area
by David_Hiers＠adp.com Sept. 14, 2011

Sept. 14, 2011

Hi, Do you get numbers from Verizon or Level3 in the DC/Alexandria VA area? I've got some delay issues in that market, looking to compare/contrast with other networks. Thanks, David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 ###Please note my email address is changing: ###from David_Hiers at adp.com ### to David.Hiers at adp.com This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

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Need some help here - Verizon Business CIC 0222 customers
by frnkblk＠iname.com Sept. 14, 2011

Sept. 14, 2011

Ah, that explains the lack of ring back. It was nice and clean for me. -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Paul Timmins Sent: Tuesday, September 13, 2011 6:40 PM To: VoiceOps at voiceops.org Subject: Re: [VoiceOps] Need some help here - Verizon Business CIC 0222 customers I did forget to mention here that the number you're calling is an asterisk echo test line over TDM. It won't ring, and if it crackles, you should probably open a ticket with your LD provider. On 09/13/2011 06:26 PM, Paul Timmins wrote: > Can anyone who is a verizon business customer on option 2 network (CIC > 0222) please call 248-574-7285 and listen quietly. If they hear static > on this call, please open a ticket with Verizon Business referencing > ticket 201109092762. > > They have a faulty DS3 into the 340 LATA interconnecting them with > AT&T's tandems and don't appear to be giving it the proper amount of > attention. Calls originating from anywhere in the nation on 0222 > should render this static. > > If you have been having unidirectional popping (heard only by the > terminating party) on calls terminating in the 248, 734, 313, 586, 810 > area codes and are a verizon business customer (or are getting calls > from verizon business customers in the 340 lata with popping), this is > why. If you have a ticket open on it, please reference mine for faster > trouble resolution. > > -Paul > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops > _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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Need some help here - Verizon Business CIC 0222 customers
by paul＠timmins.net Sept. 13, 2011

Sept. 13, 2011

Can anyone who is a verizon business customer on option 2 network (CIC 0222) please call 248-574-7285 and listen quietly. If they hear static on this call, please open a ticket with Verizon Business referencing ticket 201109092762. They have a faulty DS3 into the 340 LATA interconnecting them with AT&T's tandems and don't appear to be giving it the proper amount of attention. Calls originating from anywhere in the nation on 0222 should render this static. If you have been having unidirectional popping (heard only by the terminating party) on calls terminating in the 248, 734, 313, 586, 810 area codes and are a verizon business customer (or are getting calls from verizon business customers in the 340 lata with popping), this is why. If you have a ticket open on it, please reference mine for faster trouble resolution. -Paul

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Calls Missing GAP Information
by lists＠iamchriswallace.com Sept. 13, 2011

Sept. 13, 2011

Has anyone else seen recently a large number of calls coming into them that do not have the GAP information populated in the SS7 IAM? We see this issue come up a few times a years and we become the switchboard for people calling our customers. I know, I know? why the heck is our LRN the same as our main telephone number? Sorry, this was someone else decision years ago and we have lived with the consequences since. Yesterday morning it was Comcast, which fixed itself right before lunch time as I was getting transferred around trying to open a ticket, this morning it seems to be Verizon wireless. We have had this issue in the past with MetroPCS and Sprint, they always identify it as routing changes with phantom trunks, meaning they are doing LCR and dumping these calls wherever until an issue comes up. This in turn makes us look bad to our customers because their customers can't call them. How are other people dealing with these issues other than just constantly opening tickets, etc.? Verizon Call (This morning "Bad") ISUP message received: --------------------------------------- CIC: 1171 10010011 :CIC LSB(147) 00000100 :CIC MSB(4) --------------------------------------- --------------------------------------- Message Type: Initial address 00000001 :MESSAGE TYPE(Initial address) --------------------------------------- --------------------------------------- Nature of Connection Indicator: ......00 :SATELLITE INDICATOR(No Satellite) ....00.. :CONT. CHECK INDICATOR(Cont. Check Not Required) ...1.... :RESERVED(Outgoing Echo Control Included) 000..... :SPARE(0) --------------------------------------- --------------------------------------- Forward Call Indicators: .......0 :INCOMING_INTL_CALL(Not an Incoming International Call) .....00. :END-TO-END METHOD(No End-to-end method available) ....0... :INTERWORKING(No interworking encountered) ...0.... :SPARE1(0) ..1..... :ISUP (ISDN User Part used all the way) 01...... :ISUP PREFERENCE(ISDN User Part not required all the way) .......0 :ISDN ACCESS(Originating access non-ISDN) .....00. :SCCP METHOD(No Indication) ....0... :SPARE2(0) ...1.... :PORTED NUMBER TRANSLATION(Number translated) ..0..... :QUERY ON RELEASE(No QoR routing attempt in progress) 00...... :RESERVED(0) --------------------------------------- --------------------------------------- Calling Party's Category: 00001010 :CALLING PARTY CATEGORY(Ordinary Calling Subscriber) --------------------------------------- --------------------------------------- Called Number: 239333XXXX .0000011 :NATURE OF ADDRESS(National Number) 0....... :ODD/EVEN(Even) ....0000 :RESERVED(0) .001.... :NUMBERING PLAN(ISDN) 0....... :SPARE(0) --------------------------------------- --------------------------------------- User Service Information: ...00000 :INFORMATION TRANSFER CAPABILITY(0) .00..... :CODING STANDARD(0) 1....... :EXT1(1) ...10000 :INFORMATION TRANSFER RATE(16) .00..... :TRANSFER MODE(0) 1....... :EXT2(1) ...00010 :USER INFORMATION LAYER 1(2) .01..... :LAYER 1 IDENTIFICATION(1) 1....... :EXT3(1) --------------------------------------- --------------------------------------- OPTIONAL PARAMETERS: --------------------------------------- --------------------------------------- Calling Number: 860434XXXX .0000011 :NATURE OF ADDRESS(National Number) 0....... :ODD/EVEN(Even) ......01 :SCREENING(User Provided, screening passed) ....00.. :PRESENTATION(Presentation allowed) .001.... :NUMBERING PLAN(ISDN) 0....... :SPARE(0) --------------------------------------- --------------------------------------- Jurisdiction Information: 860434 --------------------------------------- END OF OPTIONAL PARAMETERS --------------------------------------- HEX DUMP: 9304011060100a03060d038090a20703 1032393300000a0703116840439536c4 0368404300 Same call yesterday that routed correctly? ISUP message received: --------------------------------------- CIC: 1058 00100010 :CIC LSB(34) 00000100 :CIC MSB(4) --------------------------------------- --------------------------------------- Message Type: Initial address 00000001 :MESSAGE TYPE(Initial address) --------------------------------------- --------------------------------------- Nature of Connection Indicator: ......00 :SATELLITE INDICATOR(No Satellite) ....00.. :CONT. CHECK INDICATOR(Cont. Check Not Required) ...1.... :RESERVED(Outgoing Echo Control Included) 000..... :SPARE(0) --------------------------------------- --------------------------------------- Forward Call Indicators: .......0 :INCOMING_INTL_CALL(Not an Incoming International Call) .....00. :END-TO-END METHOD(No End-to-end method available) ....0... :INTERWORKING(No interworking encountered) ...0.... :SPARE1(0) ..1..... :ISUP (ISDN User Part used all the way) 01...... :ISUP PREFERENCE(ISDN User Part not required all the way) .......0 :ISDN ACCESS(Originating access non-ISDN) .....00. :SCCP METHOD(No Indication) ....0... :SPARE2(0) ...1.... :PORTED NUMBER TRANSLATION(Number translated) ..0..... :QUERY ON RELEASE(No QoR routing attempt in progress) 00...... :RESERVED(0) --------------------------------------- --------------------------------------- Calling Party's Category: 00001010 :CALLING PARTY CATEGORY(Ordinary Calling Subscriber) --------------------------------------- --------------------------------------- Called Number: 239333XXXX .0000011 :NATURE OF ADDRESS(National Number) 0....... :ODD/EVEN(Even) ....0000 :RESERVED(0) .001.... :NUMBERING PLAN(ISDN) 0....... :SPARE(0) --------------------------------------- --------------------------------------- User Service Information: ...00000 :INFORMATION TRANSFER CAPABILITY(0) .00..... :CODING STANDARD(0) 1....... :EXT1(1) ...10000 :INFORMATION TRANSFER RATE(16) .00..... :TRANSFER MODE(0) 1....... :EXT2(1) ...00010 :USER INFORMATION LAYER 1(2) .01..... :LAYER 1 IDENTIFICATION(1) 1....... :EXT3(1) --------------------------------------- --------------------------------------- OPTIONAL PARAMETERS: --------------------------------------- --------------------------------------- Calling Number: 860434XXXX .0000011 :NATURE OF ADDRESS(National Number) 0....... :ODD/EVEN(Even) ......01 :SCREENING(User Provided, screening passed) ....00.. :PRESENTATION(Presentation allowed) .001.... :NUMBERING PLAN(ISDN) 0....... :SPARE(0) --------------------------------------- --------------------------------------- Optional Parameter: Type: 199(UNKNOWN) Length (Bits):8 00100000 :OCTET 0(32 ) --------------------------------------- --------------------------------------- Generic Address: 239325XXXX 11000000 :Type of Address(Ported Number) .0000011 :NATURE OF ADDRESS(National Number) 0....... :ODD/EVEN(Even) ......00 :RESERVED(0) ....00.. :PRESENTATION(Presentation allowed) .001.... :NUMBERING PLAN(ISDN) 0....... :SPARE(0) --------------------------------------- --------------------------------------- Jurisdiction Information: 860434 --------------------------------------- END OF OPTIONAL PARAMETERS --------------------------------------- HEX DUMP: 2204011060100a03060d038090a20703 1032393300000a0703116840439536c7 0120c008c003103239520500c4036840 4300 ---Chris

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