Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP
If I had just two such users, I would be fine. But having 10-20 or even more? It is not a nice, scalable solution... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Thursday, May 26, 2016 11:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP How about running a second asterisk instance on the same box with different IP/Port combo -- _ -- 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] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP
How about running a second asterisk instance on the same box with different IP/Port combo -- _ -- 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] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP
Hi, Unfortunately this is not the case. For some reasons (separation, fraud detection) they separate the customer using their source IP address. So I am still looking for some solution. Thanks, Attila From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Glenn Geller (VDOPh) Sent: Wednesday, May 25, 2016 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP Hi, Usually, the trunk provider(s) provide a mechanism to support this, and it's the "Tech Prefix" or just "Prefix". So, for Tenant 1, send 12348005551212 and for Tenant 2, send 56788005551212. They'll then strip the prefix, and send along to 8005551212 Most trunk providers worth anything will support this type of termination. Hope this helps, Glenn @ VDO On Wed, May 25, 2016 at 2:13 PM, Attila Megyeri mailto:amegy...@minerva-soft.com>> wrote: Hi! I would like to reopen a discussion that I saw a couple of years ago, with the subject “Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine” The use case is simpe: There are providers that want to see a separate source IP address for each of their customers SIP trunks. Now, if we have an asterisk box with several customers, we have a problem. Does anyone have experience in this topic? How could we send outgoing calls (to the same destination IP) from different source IPs depending on the caller ID (Based on From: field, sip account, preferred-identity, whatever). I was thinking about some Kamailio, or SBC that would take the calls from asterisk using a user/pass authentication on a single interface, and initiate calls from a dedicated IP address for each customer. Any better idea? Thanks Attila -- _ -- 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] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?
On Wed, May 18, 2016 at 9:44 AM, Olivier wrote: > I've got the following setup: > > PSTN ITSP SDSL Modem-Router Gateway - > Asterisk with B410P --- SIP Phones Wow. > Both SDSL Modem-Router and Gateway are managed by my ITSP. > > Some calls coming from PSTN and forwarded to an other PSTN number have a > poor voice quality. How are you forwarding them? Is it in such a way that you remain in the audio path, or do you get out of the audio path in the forward? > How can I best illustrate this ? It depends on what let has the bad audio. If it's on the SIP side (RTP to RTP) a pcap file will show you your perspective of audio losses. Received RTCP reports should show you the other side's perspective of audio losses as well. > A friend advised me to simply record incoming DAHDI channel, for instance. > How can I then translate record WAV file into meaningful figures ? If DAHDI is still in the picture in the forward scenario, that would be another place to monitor the audio. > More generaly, what would you suggest ? Try to capture each leg (IP side, using tcpdump/wireshark) and on DAHDI using dahdi_monitor or something equivalent. Figure out if any of your legs of audio quality issues. If you don't see anything, it's something at their end. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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] Avaya Phones and Asterisk
On Fri, May 20, 2016 at 8:54 AM, Diogo Cosito wrote: > Dear gentlemen, how are you? > I wonder if anyone has experience with Avaya devices, 9608G and 9641GS > models, running on SIP and using TCP transport. > The calls work well, but the callerid only "pass" number of the extension or > external number, without the name (configured correctly in sip.conf and > testing with other devices UDP works fine, like Yealink, Eyebeam, etc), > device contacts list does not work and also the hint does not work > can anybody help me? So if you use UDP transport everythng works fine? Can you post a packet capture of this happening? Also, what version of Asterisk and your sip.conf? -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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] pjsip segfault problem
no, i added patch https://trac.pjsip.org/repos/changeset/5233 to pjproject bundled testsuite with wss calls is still running Dne 26.5.2016 v 16:11 Matt Fredrickson napsal(a): Have you tried updating to pjproject version 2.5.x? It should have the patch that you listed in your other email, which I believe should be included in that branch. Hope that helps, and best of luck. Matthew Fredrickson On Thu, May 26, 2016 at 4:11 AM, Marek Červenka wrote: hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287if (t->msg_magic == msg->hdr.magic && it was only once after 2 days. i dont know how to repeat it now :( any similiar experience? -- --- Marek Cervenka === -- _ -- 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 -- --- Marek Cervenka === -- _ -- 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] pjsip segfault problem
Have you tried updating to pjproject version 2.5.x? It should have the patch that you listed in your other email, which I believe should be included in that branch. Hope that helps, and best of luck. Matthew Fredrickson On Thu, May 26, 2016 at 4:11 AM, Marek Červenka wrote: > hi, > > after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have > problem with segfault (centos 6) > > Program terminated with signal 11, Segmentation fault. > #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, > pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, > src_addr=0xb0e47a20, src_addr_len=16) > at ../src/pjnath/stun_session.c:1287 > 1287if (t->msg_magic == msg->hdr.magic && > > > it was only once after 2 days. i dont know how to repeat it now :( > > any similiar experience? > > > -- > --- > Marek Cervenka > === > > > -- > _ > -- 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 -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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] open source pbx free
> Anyone have any experience running an open source pbx and call > center solution?Need to start a call center of 10 users and i need help > > I have already installer a server with Ubuntu Server 14.04 , E1 installed > > Please advice me how to process from here > > Regards > > Yves Many of us on this list have experience running call centers off of Asterisk, myself included. If you haven't done Asterisk before, you might want to bring in some outside help in order to smooth over the process. It isn't that you can't do it on your own, but expect there to be something of a steep learning curve. If you haven't had experience with VOIP before, you will run into issues that you didn't even know were possible, and in a call center scenario, you will have people breathing down your neck wanting things fixed/changed. The great thing about Asterisk is that if you know what you are doing, you can pretty much bend it to your will. It isn't perfect (no software is), but there have been very few requests from end users that I haven't been able to fulfill once I understood what they really wanted. Phone systems are big and scary and hard for technical people. Most non-techies don't know enough about them to even know the right questions to ask. That's why your very first job is to find out what does the client really want/need their phone system to do. Call center of 10 users gives you a direction to go in, but it isn't enough to design the phone system. You need to find out what exactly do they want to happen when a call comes in. How should it be routed. Are they going to use call queues? By indicating a call center it is likely they will, but I have seen it where they don't. Once you have your requirements mostly decided, then you can go ahead and decide on what to do next. If it will fit the bill, especially for a new asterisk user, there are many prebuilt distributions that will make setting up and maintaining your Asterisk solution easier. They have nice web interfaces to handle all the heavy lifting. As you sound pretty new to VOIP, this may be the way you want to go. If they don't meet your needs, then you may be into custom programming the dialplan and gets a lot more involved. Good luck and enjoy the journey. Also, the more specific you can make your questions, the better and more likely the fine folks on this board will be to respond with helpful information. __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] open source pbx free
Anyone have any experience running an open source pbx and call center solution?Need to start a call center of 10 users and i need help I have already installer a server with Ubuntu Server 14.04 , E1 installed Please advice me how to process from here Regards Yves -- _ -- 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] pjsip segfault problem
it looks like i faced this problem https://trac.pjsip.org/repos/changeset/5233 https://issues.asterisk.org/jira/browse/ASTERISK-25275 Dne 26.5.2016 v 11:11 Marek Červenka napsal(a): hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287if (t->msg_magic == msg->hdr.magic && it was only once after 2 days. i dont know how to repeat it now :( any similiar experience? -- --- Marek Cervenka === -- --- Marek Cervenka === -- _ -- 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] pjsip segfault problem
hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287if (t->msg_magic == msg->hdr.magic && it was only once after 2 days. i dont know how to repeat it now :( any similiar experience? -- --- Marek Cervenka === -- _ -- 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
