search for: from_domain

...he DID at ipitimi ip address and caller id is DID at customer IP address. When caller id is present, the From must be the caller id number at ipitimi ip address and caller id is DID at customer IP address. The P-Asserted-Identity must be the DID at ipitimi ip address. For the endpoint, I have... from_domain = ipitimi ip address from_user = DID send_pai = yes If no caller id is present, calls go through IPitimi to my cell phone. However, if caller id is present, the P-Asserted-Identity is the caller id. Based on conversations with IPitimi and some other SIP products, this is incorrect. The P-Assert...

Hi, In PJSIP configuration, I thought that "from_domain" parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain. I've tested at the transport level, I see no changes. I've also tested with realm parameter in auth configuration, it seems to change only the digest auth value during registration...

...ge.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best Regards, > Madushan > > On Thu, Sep 15, 2016 at 8:13 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga < >> mgliyanage.rc at gmail.com> wrote: >> >>> H...

Hey guys and gals, I have a snippet of code: Net::SMTP(@host, @port, @from_domain) do |smtp| @emails.each do |email| begin smtp.send_message email.encoded, email.from, email.destinations @emails_sent += 1 rescue Exception => e # blah end end end What I want to do is: Say there are 4 emails. First email is sent OK On the second email smtp r...

...at is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >> external_signaling_address=XX.XX.XX.XX >> >> but asterisk 13 writes local ip to the from header. any suggestions? >> > Setting 'from_domain' on the endpoint will do it. Are you having issues with an internal address being used in the "From" header? > >> Best Regards, >> Madushan >> >> >> > > -- > Sent from Gmail Mobile > > -- > ________________________________________...

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

...;>>> tried >>>> >>>> external_media_address=XX.XX.XX.XX >>>> external_signaling_address=XX.XX.XX.XX >>>> >>>> but asterisk 13 writes local ip to the from header. any suggestions? >>>> >>> >> Setting 'from_domain' on the endpoint will do it. Are you having issues >> with an internal address being used in the "From" header? >> >> >> >>> >>>> Best Regards, >>>> Madushan >>>> >>>> >>>> >>> >&...

...e status of the contact (or endpoint/trunk) through AMI? Simple config: [srv_dev] type=auth auth_type=userpass username=login password=secret [srv_dev] type=aor contact=sip:sip.example.com:5060 qualify_frequency=5 default_expiration=10 max_contacts=1 remove_existing=yes [srv_dev] type=endpoint from_domain=example.com aors=srv_dev outbound_auth=srv_dev rewrite_contact=yes allow=!all,alaw Dmitriy Serov

...tie match = orange-obs.fr [Biv_Sortie] type=auth username = Biv_Sortie password=3314C9BA9688C2AA [Biv_Sortie] type=endpoint context = Isdn_Inbound dtmf_mode=rfc4733 disallow=all allow = g722, alaw, g729 direct_media=no trust_id_inbound = yes send_rpid=yes from_user = 75B55BTQUHSG at orange-obs.fr from_domain = orange-obs.fr language = en allow_subscribe = yes auth = Biv_Exit outbound_auth = Biv_Sortie aors = Biv_Sortie Question how can I solve this character problem "@"? 2) resolution of the orange-obs.fr DNS.  I am attaching an extract from the documentation that Orange issued in 2015 SI...

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip...

...ebase;pn-timeout=0;pn-tok==[redacted];pn-silent=1> While that seems odd to me, from a common-sense perspective, I don't have a deep enough background in the SIP protocol to decide if it's wrong or not, and if it's not, how to make Asterisk do it, as in something similar to the pjsip from_domain endpoint parameter that can be used to set the domain of the From: header for MESSAGEs to that endpoint. Any opinions, ideas or otherwise? Is the position of what should be in the RHS of the To: address even valid or MUST it always be the address of the next hop on path to delivering the MESSAGE?...

...lowroute] type = identify endpoint = flowroute match = 147.75.60.160/28, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28 [flowroute] type = auth username = 12345678 password = XXZZXXZZXXZZ [flowroute] type = endpoint context = from-trunk dtmf_mode = rfc4733 allow = !all,ulaw direct_media = no from_domain = us-west-wa.sip.flowroute.com tos_audio = ef tos_video = af41 ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with no audio after a few seconds. force_avp = yes auth = flowroute outbound_auth = flowroute aors = flowroute t38_udptl = yes t38_udptl_ec = fec [anonym...

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

...dev] > type=auth > auth_type=userpass > username=login > password=secret > > [srv_dev] > type=aor > contact=sip:sip.example.com:5060 > qualify_frequency=5 > default_expiration=10 > max_contacts=1 > remove_existing=yes > > [srv_dev] > type=endpoint > from_domain=example.com > aors=srv_dev > outbound_auth=srv_dev > rewrite_contact=yes > allow=!all,alaw > > Dmitriy Serov > -----BEGIN PGP SIGNATURE----- iEYEARECAAYFAlUA2aoACgkQuu7Rv+oOo/g8VgCfejBsC/X/47QnKrsoNjl3pewn YRIAnA/IzoxGm/2sQgKEI4gFv6TpwN9Q =Db9Z -----END PGP SIGNATURE-----

...erpass password=********** username=sonny ;realm=65.254.44.194 ;realm=gw1.sip.us [sonnyGW1] type=aor contact=sip:sonnyGW1 at 65.254.44.194:5060 ; tried also no username in URI [sonnyGW1] type=endpoint transport=transport-udp context=fromgw allow=!all,ulaw outbound_auth=sonnyGW1_auth aors=sonnyGW1 from_domain=gw1.sip.us [sonnyGW1] type=identify endpoint=sonnyGW1 match=65.254.44.194 ;; All endpoints for internal extensions follow -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150325/39c93196/attachment.h...

Ludovic Gasc wrote: > Hi, > > In PJSIP configuration, I thought that "from_domain" parameter in a > endpoint permits to have two SIP peers with the same usernames with a > different domain. > > I've tested at the transport level, I see no changes. > > I've also tested with realm parameter in auth configuration, it seems to > change only the dige...

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...stake? What logic to choose outgoing transport? [transport-udp] type=transport bind=0.0.0.0:5070 protocol=udp [srv_d228] type=endpoint language=ru rtp_symmetric=yes force_rport=yes disable_direct_media_on_nat=yes rewrite_contact=yes ice_support=yes disallow=all allow=alaw context=ext-fromservers from_domain=sipnet.ru from_user=talk37.ru aors=srv_d228 auth=srv_d228 set_var=fromDeviceId=228 set_var=fromUserId=2 outbound_auth=srv_d228 ;outbound_proxy=sip:sipnet.ru:5060 transport=transport-udp [srv_d228] type=aor qualify_frequency=30 contact=sip:sipnet.ru:5060 ;outbound_proxy=sip:sipnet.ru:5060 max_conta...

...s asterisk support this ? Any help is much appreciated. PJSIP config. [test01] type=aor contact=sip:34.221.174.202:15033 [test01] type = endpoint transport=transport-tls ice_support = yes allow=!all,ulaw,h264 sdp_session=xaccel rtcp_mux=yes aors=test01 media_encryption=sdes from_user=109643183 from_domain=xaccel.net outbound_proxy=sip:34.221.174.202:15033 direct_media=no direct_media_method=invite dtmf_mode=rfc4733 use_avpf=yes John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net&lt...

Something perhaps noteworth, since this is a multihomed system I bound the transport to 172.31.253.4:5060 I don't *think* that would cause Asterisk to use that IP in the FROM...at least it shouldn't. -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 2:58 PM To: 'Asterisk Users Mailing