Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file! kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "cU" kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [U/85] in [$cU] at [2 (0)] kamailio: 0(2371) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 530, column 34-36: Can't get from cache: $cU
Somehow it doesn't know what $cU is?
It looks like this now:
... # authentication route(AUTH);
$ru = "sip:" + $rU + "@3.3.3.3"; $tu = "sip:" + $tU + "@3.3.3.3"; $fu = "sip:" + $fU + "@2.2.2.2";
$var(contact_username) = $cU;
# Remove existing Contact header remove_hf("Contact");
# Insert new Contact header using the stored username insert_hf("Contact: sip:$var(contact_username)@2.2.2.2:5060\r\n");
# record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); ...
(formatting changed for E-Mail compatibility)
Thanks again, Markus
Am 22.09.2023 um 10:04 schrieb Shah Hussain Khattak:
You can start with the following:
# Change URI(s) $ru = "sip:" + $rU + "@3.3.3.3"; $tu = "sip:" + $tU + "@3.3.3.3"; $fu = "sip:" + $fU + "@2.2.2.2";
$var(contact_username) = $cU;
# Remove existing Contact header remove_hf("Contact");
# Insert new Contact header using the stored username insert_hf("Contact: sip:$var(contact_username)@2.2.2.2:5060\r\n"); OR # Insert new Contact header using the stored username insert_hf("Contact: sip:+61123123123@2.2.2.2:5060\r\n");
and then add the remaining modifications if needed as per your upstream carrier requirements.
Regards, Shah Hussain
*From:* Markus universe@truemetal.org *Sent:* Friday, September 22, 2023 8:58 AM *To:* sr-users@lists.kamailio.org sr-users@lists.kamailio.org *Subject:* [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted emergency situation (didn't monitor traffic growth properly and now encountering packet loss during peak times) as a drop-in replacement for an overloaded Asterisk box in a call-termination-to-upstream-carrier scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP 1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work - carrier is rejecting it. Carrier is authenticating our calls based on our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060 INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0 Record-Route: sip:2.2.2.2;lr Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8 Max-Forwards: 69 From: "61xxxxxxxxx" sip:+61xxxxxxxxx@1.1.1.1;tag=as3d75aadd To: sip:+32xxxxxxxx@2.2.2.2;user=phone Contact: sip:+61xxxxxxxxx@1.1.1.1:5060 Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 20.0.0 Date: Fri, 22 Sep 2023 00:06:50 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: sip:+61xxxxxxxxx@2.2.2.2;user=phone Content-Type: application/sdp Content-Length: 314 X-SIP: 1.1.1.1
v=0 o=root 1093000903 1093000903 IN IP4 1.1.1.1 s=Asterisk PBX 20.0.0 c=IN IP4 2.2.2.2 t=0 0 m=audio 25742 RTP/AVP 8 9 0 101 a=maxptime:150 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:25743 a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier without Kamailio in the path, and these are the differences I noticed, and probably the things I have to mimick with Kamailio in order to make it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0 should be INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: sip:+32xxxxxxxx@2.2.2.2;user=phone should be To: sip:+32xxxxxxxx@3.3.3.3;user=phone
From: "61xxxxxxxxx" sip:+61xxxxxxxxx@1.1.1.1;tag=as3d75aadd should be From: "61xxxxxxxxx" sip:+61xxxxxxxxx@2.2.2.2;tag=as3d75aadd
Contact: sip:+61xxxxxxxxx@1.1.1.1:5060 should be Contact: sip:+61xxxxxxxxx@2.2.2.2:5060
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060 should be Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1 should be o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show +-----+-----+----------+------+------+-----------+ | id | grp | ip_addr | mask | port | tag | +-----+-----+----------+------+------+-----------+ | 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv | | 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv | | 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv | | 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv | | 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv | | 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv | | 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv | | 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv | | 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv | +-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show dispatcher gateways +----+-------+------------------+-------+-------+------------+------+ | id | setid | destination | flags | prio. | attrs | desc | +----+-------+------------------+-------+-------+------------+------+ | 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | | +----+-------+------------------+-------+-------+------------+------+ (output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the resources to learn it as fast as I must to avoid further packet loss. If there's anyone available who can help me to get this done today, optionally in exchange for money, I'd be grateful.
Thank you! Markus __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: