[Freeswitch-dev] Consultancy?
We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis. Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements? Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX (there are also proxies and FS-based SBCs (Kamailio)) At high-level, the issues are: - Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces. Thank you Kind regards, Andy
Freeswitch has very large user community accessible through its mailing lists as well as official consultancy services provided by its original developers at signalwire.com. I, myself have been working in various Freeswitch based products and services for over 12 years and can assist if you prefer paid consultancy services. Generally speaking, if time & confidentiality permits, you can share your technical problems in user mailing list and get almost instant response from multiple experts for free which offers much more flexible solutions then hiring a single paid expert. For example, for RTP issue you mentioned, can you share sip trace at problematic endpoint (customer PABX in this case)? Thank you. -- Muhammad Shahzad Shafi Burraq Technologies UG Frankfurt, Germany Tel: +49 176 24 73 85 20 On Mon, 29 Jun 2020, 19:59 Andy Newlands, <andynewlands@gmail.com> wrote:
We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis.
Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements?
Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX (there are also proxies and FS-based SBCs (Kamailio))
At high-level, the issues are:
- Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces.
Thank you
Kind regards,
Andy _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
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Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev https://freeswitch.com
Hey Andy, not interested in the consulting myself, but as I have a small bit of using freeswitch as an sbc in the past, did you try to enable the option on your call to rewrite the rtp-timestamps? This may fix the first issue, depending on the exact sip-scenario. <param name="*rtp*-*rewrite*-*timestamps*" value="true"/> ... You can do a lot with transfers in freeswitch. If it's a refer, you'll have to implement the refer-dialplan-part, but it has been a while and maybe I'm out-of-date :). Just a quick idea that help you along maybe. Kind regards, Michem On Mon, Jun 29, 2020 at 8:44 PM Andy Newlands <andynewlands@gmail.com> wrote:
We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis.
Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements?
Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX (there are also proxies and FS-based SBCs (Kamailio))
At high-level, the issues are:
- Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces.
Thank you
Kind regards,
Andy _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev https://freeswitch.com
You should avoid setting anything that rewrites timestamps, it will preserve the jitter perfectly into the rtp flow. On Mon, Jun 29, 2020 at 3:36 PM Michel Brabants <michel.brabants@gmail.com> wrote:
Hey Andy,
not interested in the consulting myself, but as I have a small bit of using freeswitch as an sbc in the past, did you try to enable the option on your call to rewrite the rtp-timestamps? This may fix the first issue, depending on the exact sip-scenario.
<param name="*rtp*-*rewrite*-*timestamps*" value="true"/> ...
You can do a lot with transfers in freeswitch. If it's a refer, you'll have to implement the refer-dialplan-part, but it has been a while and maybe I'm out-of-date :).
Just a quick idea that help you along maybe.
Kind regards,
Michem
On Mon, Jun 29, 2020 at 8:44 PM Andy Newlands <andynewlands@gmail.com> wrote:
We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis.
Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements?
Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX (there are also proxies and FS-based SBCs (Kamailio))
At high-level, the issues are:
- Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces.
Thank you
Kind regards,
Andy _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev https://freeswitch.com
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
Hi Andy You would need to do wireshark captures trace with signalling and RTP which component FS or PBX have the issue. For FS whether you are using proxy Media or by pass media on sip profile / dial plan. Which version of FS you using, is it release version or custom build. There are so many factors to see. You can raise the issues in github as well on Freeswitch as there is a large active community. At my work we use custom build FS as SBC and its being quite stable overtime. Many thanks RegardsMuhammad Adil M Zaka+44(0)7776114280 On Monday, June 29, 2020, 06:59:06 PM GMT+1, Andy Newlands <andynewlands@gmail.com> wrote: We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis. Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements? Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX(there are also proxies and FS-based SBCs (Kamailio)) At high-level, the issues are: - Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces. Thank you Kind regards, Andy_________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev https://freeswitch.com
participants (5)
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Adil Mafzool Zaka
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Andy Newlands
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Brian West
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M S
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Michel Brabants