Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
I agree with Chad on this one. FreePBX is for a different kind of installs from what you described. I believe if you pitch it as your solution; as you have stated you will be disappointing. Before you start offering it up to the bosses as the silver bullet. I would do a test install on some spare hardware (You can use VM Ware or Hyper-V if you know what you are doing). Other wise you might be the one being shot with the silver bullet. We have used free PBX and it works great as long as your install needs are standard, but if you go beyond the standard you need something more. We are a sip trunk provider and hosted PBX provider, and we have wholesalers that try to use FreePBX to deliver customer sip trunks and they have issues.. It is the square peg round hole issue. Get the right shape peg for your communications hole.. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Chad Wallace cwall...@lodgingcompany.com Sent: Tuesday, May 08, 2012 8:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture. On Mon, 07 May 2012 06:40:46 -0600 Nunya Biznatch aster...@ihearbanjos.com wrote: I need to plan to use FreePBX on all Asterisk Servers, but I don't intend to install it until I'm in regular MAC maintenance mode. It is ashame you are going this far with your setup to rely on FreePBX. For something this complex, you are setting your self up for some heartache. It is my intention to do everything from the command line. However, there will be times when I'll have Interns coming in and doing some of the MAC activities, and I thought this might be an easier way for the day to day to get done. I've never seen it myself either, so am curious. Finally, there's the glitter factor. When my bosses come in and want a dog and pony show on the new phone system, they want to see fluffy bunnies and kittens, not the Ox that's doing the pulling. CLI = old in the minds of those that don't comprehend. I just installed FreePBX, and I'm pretty sure that do everything from the command line and use FreePBX are mutually exclusive situations. When you install FreePBX, it replaces (overwrites!) your config, and then you manually enter everything (devices, queues, IVR, etc.) in the GUI--or use a bulk import/export tool if there is one. It takes over your Asterisk install, and you have to adapt to it, not the other way around. If you want to set it up with standard Asterisk configs and your own dialplan, you'll need to find another way to do the GUI for the noobs/interns. For the bosses, I would suggest sending them to an actual dog and pony show instead. But that's just me. ;-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls through that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely omitted). Nothing seems to be wrong with that to me: v=0 o=CSM 0 1 IN IP4 x.x.x.x s=Acme c=IN IP4 x.x.x.x t=0 0 m=audio 22152 RTP/AVP 8 0 18 4 101 a=rtpmap:101 telephone-event/8000 And here's the debugging: [May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP to Off [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP o=CSM 0 1 IN IP4 x.x.x.x... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP s=Acme... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'x.x.x.x' into... [May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'x.x.x.x' and port ''. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP c=IN IP4 x.x.x.x... OK. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: chan_sip.c:9110 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0 [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Thanks, Ricardo. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Do a sip show peer PEERNAME and check the codecs allowed for that specific peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, May 09, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11 That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what peer is being selected. Andres Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.uk mailto:asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] British Telecom ISDN BRI line issues
Should I understand that no Asterisk user has issues with ISDN system access configuration from UK? or maybe no one is using Asterisk In UK :) ? On Tue, May 8, 2012 at 12:46 PM, khalid touati khalidtou...@gmail.comwrote: Hi All, I am posting this thread with the hope that someone in UK (or elsewhere) had a similar issue: Our issue is simple, we cannnot use our ISDN line, when watching asterisk console it gives a bunch of ISDN errors where the following is probably the most relevant: Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. I can post configuration and/or further information if needed. -- Khalid Touati Network Administrator CCNA -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Problem SOLVED. You'r right, this is a problem of codec mismatching. Activating sip debug i can see it: Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d (g723|ulaw|alaw|g729) [May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! I solved the problem thanks to your help! Since that SIP trunk isn't authenticated, i just receive calls in the default context that is set in sip.conf, and so, I don't set the codecs to be used. I discovered that the problem was that i had one other peer defined in sip.conf that had the same IP address set, so it was shuffling asterisk some how. Funny that the same configuration wasn't a problem in asterisk 1.4, but in this 1.8 it caused this problem. Thank you onde again, Regards, Ricardo. On Wed, May 9, 2012 at 5:10 PM, Andres and...@telesip.net wrote: On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what peer is being selected. Andres Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukmailto: asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] British Telecom ISDN BRI line issues
On 09-05-12 18:46, khalid touati wrote: Should I understand that no Asterisk user has issues with ISDN system access configuration from UK? or maybe no one is using Asterisk In UK :) ? I have no idea. But other than the error you have given very little information to go on. Which card are you using, what type of ISDN line (PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration, what do the log files say, etc? Have you tried calling the vendor of the ISDN card for support? Regards, Patrick -- _ -- 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] British Telecom ISDN BRI line issues
Thank you for your answer, I think I posted dhadi version and so but let me add more details and recap them below: We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a digium card 1HA8-0400BLF output of dahdi_hardware: pci::04:08.0 wctdm24xxp+ d161:8008 HB8- From BT side: it is called by BT a system access ISDN2 BRI (per BT NT mode and signaling as PTP) chan_dahdi.conf ; Span 1: WCBRI/0/0 HB8- (MASTER) AMI/CCS group=1,11 context= mainmenu switchtype = euroisdn signalling = bri_cpe channel = 1-2 context = default group = 63 ; Span 2: WCBRI/0/1 HB8- AMI/CCS group=1,12 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 4-5 context = default group = 63 ; Span 3: WCBRI/0/2 HB8- AMI/CCS group=1,13 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: WCBRI/0/3 HB8- AMI/CCS group=1,14 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 10-11 context = default group = 63 the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) * Thank you!! On Wed, May 9, 2012 at 1:22 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09-05-12 18:46, khalid touati wrote: Should I understand that no Asterisk user has issues with ISDN system access configuration from UK? or maybe no one is using Asterisk In UK :) ? I have no idea. But other than the error you have given very little information to go on. Which card are you using, what type of ISDN line (PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration, what do the log files say, etc? Have you tried calling the vendor of the ISDN card for support? Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC -- _ -- 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] British Telecom ISDN BRI line issues
On 09-05-12 19:54, khalid touati wrote: Thank you for your answer, I think I posted dhadi version and so but let me add more details and recap them below: We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a digium card 1HA8-0400BLF output of dahdi_hardware: pci::04:08.0 wctdm24xxp+ d161:8008 HB8- From BT side: it is called by BT a system access ISDN2 BRI (per BT NT mode and signaling as PTP) chan_dahdi.conf ; Span 1: WCBRI/0/0 HB8- (MASTER) AMI/CCS group=1,11 context= mainmenu switchtype = euroisdn signalling = bri_cpe channel = 1-2 context = default group = 63 ; Span 2: WCBRI/0/1 HB8- AMI/CCS group=1,12 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 4-5 context = default group = 63 ; Span 3: WCBRI/0/2 HB8- AMI/CCS group=1,13 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: WCBRI/0/3 HB8- AMI/CCS group=1,14 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 10-11 context = default group = 63 the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) * Thank you!! You did not provide system.conf. Do you have something like this (may have errors, I did not check): loadzone = uk defaultzone = uk span = 1,1,0,ccs,ami,te,term bchan = 1,2 hardhdlc = 3 span = 2,2,0,ccs,ami,te,term bchan = 4,5 hardhdlc = 6 span = 3,3,0,ccs,ami,te,term bchan = 7,8 hardhdlc = 9 span = 4,4,0,ccs,ami,te,term bchan = 10,11 hardhdlc = 12 Then as root: modprobe wctdm23xxp And as root: dahdi_cfg -vvv And check if all is well (green leds, happy messages in /var/log/messages, etc.). Then in chan_dahdi.conf use something like: ;BRI Module group = 1 signalling = bri_cpe context = incoming channel = 1,2,4,5,7,8,10,11 Your chan_dahdi.conf has group and context multiple times and that does not seem right (admittedly it's been ages since I setup a Digium card). Hope this helps. If not follow the installation manual step for step or call Digium support. http://docs.digium.com/H8/hx8_series_manual.pdf Regards, Patrick -- _ -- 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] British Telecom ISDN BRI line issues
Yeah sorry for that, I realized something is missing after I sent the email, but it is exactly what I have (other than order here, which doesn't really matter: you posted ami,te,term, I have ami,term,te). Actually I had couple technicians from digium look at it and they said BT equipements is not responding to the card within a certain range that the card is looking for (i'm not sure what range but I do believe too it's a BT issue), But I have run all the couple command that Patrick suggested (to double check), tested again and still same kind of errors. But Thank you very much Patrick for the guide, I was looking for that it's been a couple days!! I just hope someone that has the exact same issue or someone with previous BT experience see this and help :) ..we never know :) ! On Wed, May 9, 2012 at 2:40 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09-05-12 19:54, khalid touati wrote: Thank you for your answer, I think I posted dhadi version and so but let me add more details and recap them below: We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a digium card 1HA8-0400BLF output of dahdi_hardware: pci::04:08.0 wctdm24xxp+ d161:8008 HB8- From BT side: it is called by BT a system access ISDN2 BRI (per BT NT mode and signaling as PTP) chan_dahdi.conf ; Span 1: WCBRI/0/0 HB8- (MASTER) AMI/CCS group=1,11 context= mainmenu switchtype = euroisdn signalling = bri_cpe channel = 1-2 context = default group = 63 ; Span 2: WCBRI/0/1 HB8- AMI/CCS group=1,12 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 4-5 context = default group = 63 ; Span 3: WCBRI/0/2 HB8- AMI/CCS group=1,13 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: WCBRI/0/3 HB8- AMI/CCS group=1,14 context=mainmenu switchtype = euroisdn signalling = bri_cpe channel = 10-11 context = default group = 63 the error when placing a call : *Span: 4 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) * Thank you!! You did not provide system.conf. Do you have something like this (may have errors, I did not check): loadzone = uk defaultzone = uk span = 1,1,0,ccs,ami,te,term bchan = 1,2 hardhdlc = 3 span = 2,2,0,ccs,ami,te,term bchan = 4,5 hardhdlc = 6 span = 3,3,0,ccs,ami,te,term bchan = 7,8 hardhdlc = 9 span = 4,4,0,ccs,ami,te,term bchan = 10,11 hardhdlc = 12 Then as root: modprobe wctdm23xxp And as root: dahdi_cfg -vvv And check if all is well (green leds, happy messages in /var/log/messages, etc.). Then in chan_dahdi.conf use something like: ;BRI Module group = 1 signalling = bri_cpe context = incoming channel = 1,2,4,5,7,8,10,11 Your chan_dahdi.conf has group and context multiple times and that does not seem right (admittedly it's been ages since I setup a Digium card). Hope this helps. If not follow the installation manual step for step or call Digium support. http://docs.digium.com/H8/hx8_series_manual.pdf Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] British Telecom ISDN BRI line issues
On 09-05-12 20:57, khalid touati wrote: Yeah sorry for that, I realized something is missing after I sent the email, but it is exactly what I have (other than order here, which doesn't really matter: you posted ami,te,term, I have ami,term,te). Actually I had couple technicians from digium look at it and they said BT equipements is not responding to the card within a certain range that the card is looking for (i'm not sure what range but I do believe too it's a BT issue), But I have run all the couple command that Patrick suggested (to double check), tested again and still same kind of errors. But Thank you very much Patrick for the guide, I was looking for that it's been a couple days!! I just hope someone that has the exact same issue or someone with previous BT experience see this and help :) ..we never know :) ! Too bad you could not (yet) make it work. Hope you get somewhere with BT. Once you get past the people following those silly scripts you should be able to talk to someone who has a clue and resolve this issue. Good luck! Regards, Patrick -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
On 05/09/2012 12:59 PM, Bart Coninckx wrote: Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? That is the supported method to use in standard Asterisk, yes. DAHDI, libpri and Asterisk working together. -- 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] British Telecom ISDN BRI line issues
Yeah they have a wonderful policy that says ISDN team are not contactable :( thanks a lot!! On Wed, May 9, 2012 at 3:06 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09-05-12 20:57, khalid touati wrote: Yeah sorry for that, I realized something is missing after I sent the email, but it is exactly what I have (other than order here, which doesn't really matter: you posted ami,te,term, I have ami,term,te). Actually I had couple technicians from digium look at it and they said BT equipements is not responding to the card within a certain range that the card is looking for (i'm not sure what range but I do believe too it's a BT issue), But I have run all the couple command that Patrick suggested (to double check), tested again and still same kind of errors. But Thank you very much Patrick for the guide, I was looking for that it's been a couple days!! I just hope someone that has the exact same issue or someone with previous BT experience see this and help :) ..we never know :) ! Too bad you could not (yet) make it work. Hope you get somewhere with BT. Once you get past the people following those silly scripts you should be able to talk to someone who has a clue and resolve this issue. Good luck! Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi Bart, here is a working configuration in Netherlands: /etc/dahdi/system.conf: span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 span = 2,1,0,ccs,ami bchan = 4,5 hardhdlc = 6 span = 3,1,0,ccs,ami bchan = 7,8 hardhdlc = 9 span = 4,1,0,ccs,ami bchan = 10,11 hardhdlc = 12 loadzone= nl defaultzone= nl (of course change those to your country initials) /etc/asterisk/chan_dahdi.conf: group = 1 signalling = bri_cpe_ptmp switchtype = euroisdn context = mainmenu echocancel = yes channel = 1,2,4,5,7,8,10,11 I am not using dahdi-channels, hope it helps! On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx bart.conin...@telenet.bewrote: Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi Khalid, my setup is almost identical except for loadzone = be defaultzone = be (obviously) and in chan_dahdi.conf: [isdn4] signaling = bri_cpe_ptmp switchtype = euroisdn group = 2 context = isdn dahdichan = 10,11 this results into: ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 2(Assign awaiting TEI)! and after trying to call: May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4 [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 10: Red Alarm [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 11: Red Alarm [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 4 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 10 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 11 An old Asterisk 1.4 installation with mISDN connected to the same line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on the console however (every second). This is why I wondered if DAHDI is supposed to work over here, cheers, BC On 05/09/12 21:14, khalid touati wrote: Hi Bart, here is a working configuration in Netherlands: /etc/dahdi/system.conf: span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 span = 2,1,0,ccs,ami bchan = 4,5 hardhdlc = 6 span = 3,1,0,ccs,ami bchan = 7,8 hardhdlc = 9 span = 4,1,0,ccs,ami bchan = 10,11 hardhdlc = 12 loadzone= nl defaultzone= nl (of course change those to your country initials) /etc/asterisk/chan_dahdi.conf: group = 1 signalling = bri_cpe_ptmp switchtype = euroisdn context = mainmenu echocancel = yes channel = 1,2,4,5,7,8,10,11 I am not using dahdi-channels, hope it helps! On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi Khalid, my setup is almost identical except for loadzone = be defaultzone = be (obviously) and in chan_dahdi.conf: [isdn4] signaling = bri_cpe_ptmp switchtype = euroisdn group = 2 context = isdn dahdichan = 10,11 this results into: ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 2(Assign awaiting TEI)! and after trying to call: May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4 [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 10: Red Alarm [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 11: Red Alarm [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 4 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 10 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 11 An old Asterisk 1.4 installation with mISDN connected to the same line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on the console however (every second). bri_cpe_ptp is not a valid value for the signaling parameter. From chan_dahdi.conf.sample: ; bri_cpe:BRI PTP signalling, CPE side ; bri_net:BRI PTP signalling, Network side ; bri_cpe_ptmp: BRI PTMP signalling, CPE side ; bri_net_ptmp: BRI PTMP signalling, Network side Richard -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Right you are, but when using bri_cpe I get: [May 9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI Error on span 4: Received MDL/TEI managemement message, but configured for mode other than PTMP! This repeats itself every second. The bri_cpe_ptmp settings seems to give the least troubles, but no calling possible, BC On 05/09/12 22:10, Richard Mudgett wrote: Hi Khalid, my setup is almost identical except for loadzone = be defaultzone = be (obviously) and in chan_dahdi.conf: [isdn4] signaling = bri_cpe_ptmp switchtype = euroisdn group = 2 context = isdn dahdichan = 10,11 this results into: ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 2(Assign awaiting TEI)! and after trying to call: May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4 [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 10: Red Alarm [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 11: Red Alarm [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 4 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 10 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 11 An old Asterisk 1.4 installation with mISDN connected to the same line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on the console however (every second). bri_cpe_ptp is not a valid value for the signaling parameter. From chan_dahdi.conf.sample: ; bri_cpe:BRI PTP signalling, CPE side ; bri_net:BRI PTP signalling, Network side ; bri_cpe_ptmp: BRI PTMP signalling, CPE side ; bri_net_ptmp: BRI PTMP signalling, Network side Richard -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Please just reply to the mailing list. - Original Message - Right you are, but when using bri_cpe I get: [May 9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI Error on span 4: Received MDL/TEI managemement message, but configured for mode other than PTMP! This repeats itself every second. The bri_cpe_ptmp settings seems to give the least troubles, but no calling possible, Are you able to make calls when in PTP mode? The warning message is just complaining about receiving unexpected TEI management messages because the span is in PTP mode. It is otherwise benign if the line is really PTP. If you can make calls, please create a JIRA issue on the PRI project so the message level can be reduced. Please attach an intense pri debug output showing the received MDL messages. pri set debug 2 span 4 https://issues.asterisk.org/jira Richard -- _ -- 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] Broadvoice Got SIP response 503 Service Unavailable
The solution: Stop the Asterisk, made a backup of configuration files, delete files and modules previously installed. Then Reinstall with default config files with make samples and I only insert the configuration of Broadvoice and immediately worked. Then I restore the configuration files /etc/asterisk (thinking it was a problem of modules) and stopped working again, It mean that the problem was in the configuration files, or too much or too less lines. So what I made is transfer line x line what I needed from old to new configuration file. As soon detected where it was what was missing or left over, I will publish the list regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe El vie, 04-05-2012 a las 09:23 +, isr...@gmail.com escribió: Broadvoice has a lot of problems for the last 2 months -Original Message- From: Ing. CIP Alejandro Celi Mariategui a...@linux.org.pe Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 04 May 2012 02:11:11 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable Hi, I'm running Asterisk 1.8.11.1 @office. The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working. No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems Any suggestions or thoughts? Alex Celi This is the info central*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status 488/488181.64.96.122D 11037OK (182 ms) sip.broadvoice.com/305422 206.15.148.221 5060 OK (131 ms) sip.conf externip=190.12.68.20 localnet=192.168.20.0/255.255.255.0 localnet=192.168.10.0/255.255.255.0 nat=comedia pedantic=no register = 3054221...@sip.broadvoice.com:XX:3054221...@sip.broadvoice.com [sip.broadvoice.com] type=friend host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3054221494 defaultuser=3054221494 authname=3054221494 secret=X context=entrantes dtmfmode=inband dtmf=inband nat=comedia directmedia=no qualify=yes callgroup=1 pickupgroup=1 disallow=all allow=ulaw allow=alaw I turned on sip debug. This is what I received 181.64.96.122: Is my home IP 190.12.68.20 or central.cipher.pe: is office IP 206.15.148.221: Broadvoice Server --- SIP read from UDP:181.64.96.122:11037 --- INVITE sip:90018006273...@central.cipher.pe SIP/2.0 Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport Max-Forwards: 70 Contact: sip:488@181.64.96.122:11037 To: 90018006273999sip:90018006273...@central.cipher.pe From: 488sip:4...@central.cipher.pe;tag=93cce179 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 56015 Content-Length: 235 v=0 o=- 8 2 IN IP4 192.168.7.33 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.7.33 t=0 0 m=audio 2424 RTP/AVP 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424 a=sendrecv - --- (12 headers 10 lines) --- Sending to 181.64.96.122:11037 (NAT) Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. Found peer '488' for '488' from 181.64.96.122:11037 --- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;received=181.64.96.122;rport=11037 From: 488sip:4...@central.cipher.pe;tag=93cce179 To: 90018006273999sip:90018006273...@central.cipher.pe;tag=as77d2f824 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 1 INVITE Server: Asterisk PBX 1.8.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0a1fded4 Content-Length: 0 -- _ -- 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] Why do I get call twice in one go?
I understand why do I get call twice to my mobile when I execute the following AMI command sets: ACTION: Originate Channel: Local/800@test Timeout: 6 Priority: 1 and my dialplan look like this: [test] exten = 800,1,DIAL(SIP/447xx@voip); exten = 800,n,Hangup() How to prevent getting called twice in one go when I execute this AMI command? Thanks... -- _ -- 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] Digium IP Phones
Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -- _ -- 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
